Re: [asterisk-users] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes
On 10/23/2014 05:00 PM, Matthew Jordan wrote:
On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton
dfullertaster...@shorelinecontainer.com
mailto:dfullertaster...@shorelinecontainer.com wrote:
Hello all,
I'm setting up a couple of test boxes and I'm running into a
problem. What I need help with is determining whether I'm going
something wrong or if I need to post a bug report. I have two
asterisk 13.0-beta 3 machines set up with extensions connected to
each as such:
3700 AST-A -- AST-B 3800 3801
When I place a call from 3800 to 3700 or the other way around ,
asterisk seg faults on both machines at roughly the same time. All
connections are done using PJSIP. The crash occurs when the ringing
extension is answered.
If I set (directmedia=no) OR (directmedia=yes t38_udptl=yes) on
the trunk then the call completes fine. All phones and servers are
on the same LAN with no firewalls active.
The trunk between AST-A and AST-B is configured like this in
pjsip.conf and is identical on both machines:
[transport-lan]
type=transport
protocol=udp
bind=0.0.0.0
tos=af31
[pbxbeta]
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
context=phone-level3
aors=pbxbeta
send_rpid=no
send_pai=yes
trust_id_inbound=yes
trust_id_outbound=yes
direct_media=yes
direct_media_glare_mitigation=__outgoing
;direct_media_method=update
tos_audio=46
tos_video=34
t38_udptl=no
t38_udptl_nat=no
[pbxbeta]
type=aor
contact=sip:{remote IP address}:5060
[pbxbeta]
type=identify
endpoint=pbxbeta
match={remote IP address}
The phones have the following set in pjsip.conf (snippet):
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
send_rpid=no
send_pai=yes
direct_media=yes
tos_audio=46
tos_video=34
Is there something I'm doing wrong here?
Thanks
Asterisk shouldn't crash.
Please file a bug report ASAP at issues.asterisk.org
http://issues.asterisk.org, with a properly generated backtrace:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
Created: https://issues.asterisk.org/jira/browse/ASTERISK-24448
Let me know if you need any more information.
Thanks
-Dave
--
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Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
On 10/22/2014 03:55 PM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) Thanks! --Tim I can't help with your root problem (maybe check core show function FAXOPT?), but the spandsp site is up. Try using www.spandsp.org. Downloads are available here: http://www.spandsp.org/downloads/spandsp/ -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes
Hello all,
I'm setting up a couple of test boxes and I'm running into a problem.
What I need help with is determining whether I'm going something wrong
or if I need to post a bug report. I have two asterisk 13.0-beta 3
machines set up with extensions connected to each as such:
3700 AST-A -- AST-B 3800 3801
When I place a call from 3800 to 3700 or the other way around , asterisk
seg faults on both machines at roughly the same time. All connections
are done using PJSIP. The crash occurs when the ringing extension is
answered.
If I set (directmedia=no) OR (directmedia=yes t38_udptl=yes) on the
trunk then the call completes fine. All phones and servers are on the
same LAN with no firewalls active.
The trunk between AST-A and AST-B is configured like this in pjsip.conf
and is identical on both machines:
[transport-lan]
type=transport
protocol=udp
bind=0.0.0.0
tos=af31
[pbxbeta]
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
context=phone-level3
aors=pbxbeta
send_rpid=no
send_pai=yes
trust_id_inbound=yes
trust_id_outbound=yes
direct_media=yes
direct_media_glare_mitigation=outgoing
;direct_media_method=update
tos_audio=46
tos_video=34
t38_udptl=no
t38_udptl_nat=no
[pbxbeta]
type=aor
contact=sip:{remote IP address}:5060
[pbxbeta]
type=identify
endpoint=pbxbeta
match={remote IP address}
The phones have the following set in pjsip.conf (snippet):
type=endpoint
disallow=all
allow=g722
allow=ulaw
transport=transport-lan
send_rpid=no
send_pai=yes
direct_media=yes
tos_audio=46
tos_video=34
Is there something I'm doing wrong here?
Thanks
-Dave
--
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Re: [asterisk-users] Asterisk 12.4 IMAP VM Issue - Can't move messages between folders
On 07/17/2014 09:46 AM, Dave Fullerton wrote: Hello all, I'm running into an issue with Asterisk 12.4 and IMAP voicemail. I have asterisk set up to connect to my Dovecot IMAP server and I can leave and retrieve messages from my inbox and old messages. However, I am unable to move messages between folders. I get a message from asterisk stating Sorry the users mailbox can't accept more messages. Here is my setup: In Voicemail.conf I have the following set: imapgreetings=no imapparentfolder=Voicemail imapfolder=Voicemail.Inbox On my imap server, I have the following folder structure: INBOX Sent Junk Drafts Voicemail |-Family |-Inbox |-Work I did a packet capture on my imap server and found that when I go to move a message from Old messages to Family the following happens: Asterisk issues a CREATE Voicemail.Family which succeeds with OK Create completed (The folder is successfully created if it does not exist, I can see it in thunderbird). Then Asterisk issues a COPY 1 Family which fails with NO [TRYCREATE] Mailbox Doesn't exist: Family I don't think Asterisk is using the value of imapparentfolder when copying the message. The COPY command should be COPY 1 Voicemail.Family. Is there something I am missing in my configuration or is this a bug? Thank you -Dave I think I have tracked the issue down to save_to_folder in app_voicemail.c. The third argument to mail_move/mail_copy needs to be different, but my C is not strong enough to know what I need to change it to. Any suggestions? -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 12.4 IMAP VM Issue - Can't move messages between folders
Hello all, I'm running into an issue with Asterisk 12.4 and IMAP voicemail. I have asterisk set up to connect to my Dovecot IMAP server and I can leave and retrieve messages from my inbox and old messages. However, I am unable to move messages between folders. I get a message from asterisk stating Sorry the users mailbox can't accept more messages. Here is my setup: In Voicemail.conf I have the following set: imapgreetings=no imapparentfolder=Voicemail imapfolder=Voicemail.Inbox On my imap server, I have the following folder structure: INBOX Sent Junk Drafts Voicemail |-Family |-Inbox |-Work I did a packet capture on my imap server and found that when I go to move a message from Old messages to Family the following happens: Asterisk issues a CREATE Voicemail.Family which succeeds with OK Create completed (The folder is successfully created if it does not exist, I can see it in thunderbird). Then Asterisk issues a COPY 1 Family which fails with NO [TRYCREATE] Mailbox Doesn't exist: Family I don't think Asterisk is using the value of imapparentfolder when copying the message. The COPY command should be COPY 1 Voicemail.Family. Is there something I am missing in my configuration or is this a bug? Thank you -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] recommendations for RJ-11 surge supressors?
On 06/27/2013 10:37 AM, Andrew Latham wrote: On Thu, Jun 27, 2013 at 10:34 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 27 June 2013, Eric Cooper wrote: I'd like to protect my expensive Digium FXO cards from spikes on my three incoming PSTN lines. Does anyone have any recommendations? Does your telco not fit surge suppressors to the NTE as a matter of standard practice? Perhaps we are spoiled in the UK . -- AJS Answers come *after* questions. APC sells a modular solution that has rack mount or wall mount options. ProtectNet is the product line. http://www.apc.com/products/family/index.cfm?id=145 I'll second the APC option. A PRM4 and two PTEL2 will protect 4 lines with a little wiring. Make sure you have a good ground to connect to or the whole thing is worthless. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI-linux 2.7 compile error with CONFIG_DAHDI_NET enabled
Not sure how I should officially report this, but I'm getting a compile error with DAHDI-linux 2.7 when I define CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h. I am able to compile successfully when I leave it undefined, but I need to be able to use the network support. snipped /oct6100_api/oct6100_tsst.o AR /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/oct612x/lib.a Building modules, stage 2. MODPOST 0 modules make[1]: Leaving directory `/usr/src/linux-3.4.45' make -C /lib/modules/3.4.45-smp/build SUBDIRS=/tmp/dahdi-linux-2.7.0-net/drivers/dahdi DAHDI_INCLUDE=/tmp/dahdi-linux-2.7.0-net/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-3.4.45' CC [M] /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c: In function 'dahdi_net_open': /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c:1967:4: error: 'struct dahdi_chan' has no member named 'rxbufpolicy' make[2]: *** [/tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o] Error 1 make[1]: *** [_module_/tmp/dahdi-linux-2.7.0-net/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-3.4.45' make: *** [modules] Error 2 -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI-linux 2.7 compile error with CONFIG_DAHDI_NET enabled
On 06/10/2013 11:53 AM, Shaun Ruffell wrote: On Mon, Jun 10, 2013 at 11:33:16AM -0400, Dave Fullerton wrote: Not sure how I should officially report this... You should feel free to open issues at http://issues.asterisk.org. but I'm getting a compile error with DAHDI-linux 2.7 when I define CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h. I am able to compile successfully when I leave it undefined, but I need to be able to use the network support. snipped /oct6100_api/oct6100_tsst.o AR /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/oct612x/lib.a Building modules, stage 2. MODPOST 0 modules make[1]: Leaving directory `/usr/src/linux-3.4.45' make -C /lib/modules/3.4.45-smp/build SUBDIRS=/tmp/dahdi-linux-2.7.0-net/drivers/dahdi DAHDI_INCLUDE=/tmp/dahdi-linux-2.7.0-net/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-3.4.45' CC [M] /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c: In function 'dahdi_net_open': /tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.c:1967:4: error: 'struct dahdi_chan' has no member named 'rxbufpolicy' make[2]: *** [/tmp/dahdi-linux-2.7.0-net/drivers/dahdi/dahdi-base.o] Error 1 make[1]: *** [_module_/tmp/dahdi-linux-2.7.0-net/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-3.4.45' make: *** [modules] Error 2 Thanks for reporting this. I have a patch [1] for the next release. If you are willing, care to apply it to your 2.7.0 tree and check it out? If you are building from a tarball you can easily apply it like: $ curl http://git.asterisk.org/gitweb/?p=team/sruffell/dahdi-linux.git;a=patch;h=e4d89ffa7485; | patch -p1 [1] http://git.asterisk.org/gitweb/?p=team/sruffell/dahdi-linux.git;a=commitdiff;h=e4d89ffa7485 Cheers, Shaun Thank you Shaun, that patch did the trick. DAHDI compiled and appears to be functioning normally. I wondered if I might impose upon you for a question. I am in the process of replacing an old router with a T1 interface with a Linux machine. My test rig is currently using a spare TE220F. I know digium's card were primarily designed to function in a telephony role, but is there any technical reason I should not use them in an exclusively data role as well? I am trying to decide if I should purchase another TE220F (which I have experience with) or use a Sangoma product (which I do not). Thank you for your time. -- Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My new Polycom 450's can't xfer to 4-digit extension
On 05/04/2013 08:43 PM, Mike Diehl wrote: Hi all. I just installed bunch of IP450's and everything went well and my customer is happy except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries to dial and the transfer fails. I feel sure that it's a dial plan issue on the phone. We are running: PolycomSoundPointIP-SPIP_450-UA/3.3.3.0094 The dialplan section of the sip.cfg provisioning file is: dialplan dialplan.1.impossibleMatchHandling=0 dialplan.1.removeEndOfDial=1 dialplan.1.applyToUserSend=1 dialplan.1.applyToUserDial=1 dialplan.1.applyToCallListDial=0 dialplan.1.applyToDirectoryDial=0 snip digitmap dialplan.1.digitmap= dialplan.1.digitmap.timeOut= dialplan.2.digitmap= dialplan.2.digitmap.timeOut= dialplan.3.digitmap= dialplan.3.digitmap.timeOut= snip The same section of the phone1.cfg file is: dialplan dialplan.1.impossibleMatchHandling=0 dialplan.1.removeEndOfDial=1 dialplan.1.applyToUserSend=1 dialplan.1.applyToUserDial=1 dialplan.1.applyToCallListDial=0 snip dialplan.6.applyToDirectoryDial=0 digitmap dialplan.1.digitmap= dialplan.1.digitmap.timeOut= dialplan.2.digitmap= dialplan.2.digitmap.timeOut= snip /routing /dialplan The MAC-specific provisioning file does not have a dialplan section. All of my Polycom users share these files and many of them can transfer to 4-digit extensions. Is there something I need to do for the 450 to make this work? Thank you in advance. Mike Diehl. You really should configure the dialplan.digitmap attribute. I don't know why the other polycom phones are working and the 450's are not, but I believe the combination of not having a digitmap configured and having dialplan.impossibleMatchHandling set to 0 is what is causing the problem. You could try setting dialplan.impossibleMatchHandling to 2 for the short term, but configuring the digitmap to match your environment is the best solution. Check the SIP admin guide for details on how to set it up. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Soundpoint IP 330 provisioning
Daniel, The bootom is not part of the SIP application that you downloaded. You need to download the appropriate bootrom from the link Kevin supplied. Before you do any more though, you really need to download the SoundpointIP Admin guide here: http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf Read chapters 2 and 3 at a minimum. There is a lot to setting up a provisioning system for polycom phones and it helps to have the proper background before getting started. -Dave On 04/12/2013 01:50 PM, Daniel - Asterisk wrote: Hello Kevin, Could you please tell me where I can found the 'application' my phones are looking for? I've already downloaded spip_ssip_vvx_3_2_3_release_sig combined and split zips, which lack a bootrom.ld file Thank you! Elder On Fri, Apr 12, 2013 at 12:44 PM, Kevin Larsen kevin.lar...@pioneerballoon.com mailto:kevin.lar...@pioneerballoon.com wrote: _http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html_ Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: Daniel - Asterisk earohua...@gmail.com mailto:earohua...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com, Date: 04/12/2013 12:42 PM Subject: [asterisk-users] Polycom Soundpoint IP 330 provisioning Sent by: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com Hello all, I need the bootrom.ld file to set up some Polycoms I have Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A I've publiched on my FTP files downloaded from _http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html_ (3.2.3 combined and split zips) but my phones are still showing the message: error, application is not present I apologize it is not a pure Asterisk question but I'm sure some of you can help me. Thanks in advance! Elder Arohuanca Lima - Peru-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- DO NOT SEND WITH THIS ACCOUNT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Polycom IP450 Firmware Issues
On 12/06/2012 04:09 PM, Tim Nelson wrote: I have a site with Polycom handsets on all the desks, mostly IP650s, some IP550s, and some IP450s as well. I need to update the firmware on the IP450s. However, the firmware simply won't load. The latest firmware (4.0.3 Rev F) supports all phones at this site, and was downloaded from here: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html The phone pulls the firmware from the PBX via TFTP (as expected), but always results in 'Error: Image is not compatible with the phone'. As a troubleshooting step, *ALL* firmware has been removed from the TFTP root, and *ONLY* the new firmware placed there. So, is the Polycom firmware matrix wrong about this phone/firmware compatibility, or am I missing something? The bootrom has also been upgraded to the latest without any problems. Thoughts? My head is getting sore from banging it on my desk... :/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 Without knowing what version of the SIP application is already on the phones it is hard to say. But it sounds as if your phones are loaded with SIP 3.2.x or lower. To upgrade beyond 3.2.x you will need to put the 4.4 version of the BootROM on your provisioning server. With SIP 4.0 and later (and I believe 3.3 as well) the BootROM (now called the updater) is included in the sip.ld file itself. The 4.4 version of the BootROM updates the phone to look for the new updater inside the sip.ld file. It should ONLY be used to upgrade phones from SIP 3.2 or lower to SIP 3.3 or higher. Hope this helps. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0
On 07/26/2012 04:28 PM, Tim Nelson wrote: Greetings- I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0 system. Everything is running smoothly with few problems. However, I have an issue that maybe someone could shed light on... Many of the phones have 'buddy watch' enabled for the other phones, basically Polycom's version of BLF. This works fine when watched extensions are on the phone, ringing, etc, as the LED lights/flashes appropriately for the status. However, the phones also offer various presence states such as 'Out to Lunch' or 'Away from Desk' etc. When a phone is set to one of these presence states, the other phones watching never show that status. Does that make sense? Is there any reason why those states would not propagate between the phones (through Asterisk?) ? And, on a side note, if anyone knows how to remove the 'thistle' background from a Polycom phone I'd be especially delighted. It was set by a user on a device, and there is no option to remove it, or replace it with the blank background which is the default. :/ If you just want to reset the background on that phone then you want: Menu, 3, 1, 1, 4, 2, 2, Select (Seems like you should get god mode for that too, but alas, no). If you want to prohibit anyone from setting that particular background you could always remove the jpg from the provisioning server. As for the buddy status, I don't think it works (or ever will work) with asterisk. I always turn that button off when I set up my site sip.conf to avoid any questions. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware 4.0.1 and paging
Which version of asterisk are you using? I just have this in 1.4 and it works fine: SIPAddHeader(Alert-Info: intercom); -Dave On 02/14/2012 08:10 PM, Mike wrote: In case anybody was following this thread, or someone Googles it in the future, here is the solution: This worked fine with Polycom firmware 3.3x: exten = s,n,SIPAddHeader(Alert-Info:Ring Answer) For firmware 4.0+, apparently I needed to add info=, i.e.: exten = s,n,SIPAddHeader(Alert-Info: info=Ring Answer) Simple, yet quite obscure (for me at least). Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, February 13, 2012 10:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Thanks Dave, it at least gives me hope that my efforts aren`t wasted. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Monday, February 13, 2012 9:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On 02/10/2012 05:30 PM, Mike wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? I would second the others suggestions about rewriting the configs. Polycom made extensive changes between 3.2 and 3.3, and I think they made a fair number of changes between 3.3 and 4.0. I have two phones that I've upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I believe I have auto answer working as you describe. Here's the pertinent snippet from my config: polycomConfig voIpProt voIpProt.SIP voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.1.class=ringAutoAnswer voIpProt.SIP.alertInfo.1.value=intercom voIpProt.SIP.alertInfo.2.class=ringAnswerMute voIpProt.SIP.alertInfo.2.value=page voIpProt.SIP.alertInfo.3.class=autoAnswer voIpProt.SIP.alertInfo.3.value=silentanswer /voIpProt.SIP.alertInfo /voIpProt.SIP /voIpProt /polycomConfig I have also added anse.rt section to adjust the ringer and timeouts for these ring tones. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware 4.0.1 and paging
On 02/10/2012 05:30 PM, Mike wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it’s worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? I would second the others suggestions about rewriting the configs. Polycom made extensive changes between 3.2 and 3.3, and I think they made a fair number of changes between 3.3 and 4.0. I have two phones that I've upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I believe I have auto answer working as you describe. Here's the pertinent snippet from my config: polycomConfig voIpProt voIpProt.SIP voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.1.class=ringAutoAnswer voIpProt.SIP.alertInfo.1.value=intercom voIpProt.SIP.alertInfo.2.class=ringAnswerMute voIpProt.SIP.alertInfo.2.value=page voIpProt.SIP.alertInfo.3.class=autoAnswer voIpProt.SIP.alertInfo.3.value=silentanswer /voIpProt.SIP.alertInfo /voIpProt.SIP /voIpProt /polycomConfig I have also added an se.rt section to adjust the ringer and timeouts for these ring tones. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom soundpint ip650 question
Yes, her extension is now on one line key, but this does not mean she cannot make a new call if she is already on a line. If she is on a call and needs to call someone else she would put the first call on hold, then the New Call soft key will appear. You can press that to get dialtone and place an outgoing call. This setup breaks (at least, I haven't found a way around it) if set up additional registrations on the other line keys. Lets say you have extension 100 on the first line key with 6 calls per key, and extension 101 on the second key with 6 calls per key. You have two incoming calls on extension 100 and you wish to make a call. If you press the New Call soft key it gives you dialtone on extension 101 instead of on 100. I have not figured out how to fix this yet, but I almost never have phones that are registered with more than one extension (except mine of course). -Dave On 11/17/2011 11:57 AM, eherr wrote: Doing it that was does accomplish the original question, which is cool. Thanks. But you're also right in that we wont like it. This setup only allows for her extension to be registered to just one line key, unless I am missing something. So in order for her to dial out, I would need to assign the rest of the line keys a different extension and set it up as normal. Under this setup, I am assuming that internally, her coworkers will have to know both extensions that she has, right? Thanks, --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, November 17, 2011 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] polycom soundpint ip650 question No, this is all done through the phone provisioning. You are correct, you would set reg.1.lineKeys=1 but then set reg.1.callsPerLineKey=6 or 4 or whatever you want. In the default phone1.cfg callsPerLineKey immediately follows the lineKeys setting. -Dave On 11/17/2011 10:58 AM, eherr wrote: I basically understand what you're saying but I am a little confused. Are you saying.. Reg.1.lineKeys=1 Then on asterisk allow 4 calls per sip extension Thanks, --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, November 17, 2011 10:18 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] polycom soundpint ip650 question On 11/16/2011 01:06 PM, eherr wrote: On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E There is a way, but you may not like it. When you provision the phone you can specify how many line keys and how many calls per line key you want. If you specify one line key but (lets say) 4 calls per line key, then the phone will display a scrollable list of your current calls, including the caller ID. You use the up and down arrow to select which caller you want and then the soft keys to perform a function (hold, resume, transfer, end call). It can require a few more button presses than one call per line key for certain things and it doesn't work well if you have the phone set up with multiple registrations. We use this method on our phones here for the very reason. People don't seem to mind. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X86_64 Compilation Issue
On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote: Hi, compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am seeing the following when running the make: /usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.so when searching for -lcrypto /usr/bin/ld: skipping incompatible /usr/lib/libcrypto.a when searching for -lcrypto How can I get Asterisk to pick up the 64 bit version of the libraries instead of the 32 bit ones ? Is it just a case of updating LD_LIBRARY_PATH ? -- Thanks, Phil Did you run configure with --libdir=/usr/lib64 ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging with Polycom 3.3.x
Actually, I don't think that has been the case for quite a while. Anyone can get the latest firmware directly from polycom. Including, 3.3.1F http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html On 02/24/2011 03:32 PM, Mike wrote: Sorry, I realize my tone might not go down well. I didn't mean to sound like a jerk, but I was just stating that resellers are also authorized to distribute the firmware to their customers if I recall correctly, so everybody can get the firmware for free, just not directly from Polycom. And I don't actually think this is the best way for Polycom to do things, but that`s the way things are. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 24, 2011 3:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Polycom are at 3.3.1 now, so 3.3.0 should be fair game. It has nothing to do with paying or not, the company that sold you the phone should be able to give you the latest version no? Unless you bought from a guy who found a box that fell off a truck.or some third-rate reseller. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 24, 2011 3:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Is 3.3.x downloadable for non-paying people yet? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 24, 2011 2:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Paging with Polycom 3.3.x Thank you Terry and Ryan, I will try those things and see if I can find my problem. Will definitely come back with my solution in case it helps somebody else. Mike Hi Terry, I did that, and did find a autoRingAnswer and Ringanswer modes that I tried to use, but somewhere I am missing something that breaks it. If ever you find what you did, I`d appreciate if you'd share with me. Mike alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ and RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=2000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ where the timeout is the ampount of time on milliseconds before it goes to speaker. These values are in the sip.cfg, so in your server it may be sip_316.cfg. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- DO NOT SEND WITH THIS ACCOUNT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax/Modem, Asterisk, Channel Banks
On 08/03/2010 10:48 AM, Joel Maslak wrote: I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm needing a solution for fax machines that works as well as a POTS line from my carrier. If the POTS line is the solution, I'll keep it, but I'd rather move away from that. Here's what I'm thinking...will it work? I would use a dual-port Digium T1 card. In one port, I'd terminate a telco PRI T1. In the other port, I'd terminate a Rhino channel bank, connected to each of my fax machines (and a stamp machine with an internal modem). What I'm wanting is to be able to send/receive faxes via the telco PRI and the analog fax machines. I also want the stamp machine to work. I don't want this to work 98% as well as the Telco - they truly need to work 100% as well. So...will this work? It should. That's the setup I'm using (but with an Adit 600 channel bank) and it works perfectly. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Y-cords - What are they ?
On 07/08/2010 10:19 AM, Zeeshan Zakaria wrote: That's why I specifically mentioned Cat5 networks, because giga bit networks which use four pairs are called Cat6 networks. This is true that Cat5 networks are also used with gigabit hardware, but technically it is wrong. Cat6 hardware uses different frequencies over copper than Cat5, and mixing and matching Cat5 and Cat6 results in not a true gigabit performance. And certainly there are no Y-cables in Cat6 networks. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 9:55 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Zeeshan Zakariazisha...@gmail.com writes: making use of the fact that both Cat5 networks and B... For Ethernet, this is only true for 10Mbps and 100Mbps. Gigabit and up uses all four pairs. /Benny According to wikipedia: http://en.wikipedia.org/wiki/Gigabit_Ethernet Cat6 wiring is only a requirement of 1000BASE-TX equipment which only uses two pairs. 1000BASE-T, which is more common, does use all four pairs but can use Cat5 or higher wiring. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Hylafax
On 06/15/2010 12:48 PM, Samantha wrote:
Hey Guys
I have hylafax working about 95%
The problem is I have a DID for fax 0742244224
When I receive a fax I see in the log file
n 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-sip-external:1] NoOp(SIP/5060-0a2f7308, Received
incoming SIP connection from unknown peer to 0742244224) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-sip-external:2] Set(SIP/5060-0a2f7308, DID=0742244224)
in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-sip-external:3] Goto(SIP/5060-0a2f7308, s,1) in new
stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Goto
(from-sip-external,s,1)
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[...@from-sip-external:1] GotoIf(SIP/5060-0a2f7308,
1?from-trunk,0742244224,1) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Goto
(from-trunk,0742244224,1)
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:1] Set(SIP/5060-0a2f7308, __FROM_DID=0742244224)
in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:2] Gosub(SIP/5060-0a2f7308,
app-blacklist-check,s,1) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[...@app-blacklist-check:1] GotoIf(SIP/5060-0a2f7308, 0?blacklisted) in
new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[...@app-blacklist-check:2] Return(SIP/5060-0a2f7308, ) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:3] ExecIf(SIP/5060-0a2f7308, 1
?Set(CALLERID(name)=0282086500)) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:4] Set(SIP/5060-0a2f7308, FAX_RX=4111) in new
stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:5] Set(SIP/5060-0a2f7308,
fax_rx_email=s...@smellyblackdog.com.au) in new stack
[Jun 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:6] Answer(SIP/5060-0a2f7308, ) in new stack
[Jun 16 02:44:20] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:7] PlayTones(SIP/5060-0a2f7308, ring) in new
stack
[Jun 16 02:44:20] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-trunk:8] NVFaxDetect(SIP/5060-0a2f7308, 0|t) in new
stack
[Jun 16 02:44:20] DEBUG[3679] app_nv_faxdetect-1.0.6_1.4.c: Preparing detect
of fax (waitdur=4ms, sildur=1000ms, mindur=100ms, maxdur=-1ms)
[Jun 16 02:44:24] DEBUG[3679] app_nv_faxdetect-1.0.6_1.4.c: Fax detected on
SIP/5060-0a2f7308
[Jun 16 02:44:24] NOTICE[3679] app_nv_faxdetect-1.0.6_1.4.c: Redirecting
SIP/5060-0a2f7308 to fax extension
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[...@from-trunk:1] Goto(SIP/5060-0a2f7308, ext-fax,in_fax,1) in new
stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Goto (ext-fax,in_fax,1)
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[in_...@ext-fax:1] StopPlayTones(SIP/5060-0a2f7308, ) in new stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[in_...@ext-fax:2] GotoIf(SIP/5060-0a2f7308, 0?3:analog_fax,1) in new
stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Goto (ext-fax,analog_fax,1)
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[analog_...@ext-fax:1] GotoIf(SIP/5060-0a2f7308, 0?4:2) in new stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Goto (ext-fax,analog_fax,2)
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[analog_...@ext-fax:2] Set(SIP/5060-0a2f7308, DIAL=IAX2/4111) in new
stack
[Jun 16 02:44:24] VERBOSE[3679] logger.c: -- Executing
[analog_...@ext-fax:3] Dial(SIP/5060-0a2f7308,
IAX2/4111/0282086500,20,d) in new stack
My FaxDispatch config is
#!/bin/sh
##
## FaxDispatch
## (see `man faxrcvd` for moreyyy
# The numbers before the paren correspond to asterisk extensions in
# extensions.conf
case $CALLID4 in
# customer DID routing:
0742242442) sendto=...@smellyblackdog.com.au; FILETYPE=pdf;;
# everything else goes to default case:
*) sendto=...@smellyblackdog.com.au; FILETYPE=pdf;;
esac
The problem is that it ignores the called number in the did and drops
through to the default
I have also done the relevant mod to the /etc/asterisk/extensions.conf file
as well
Any Ideas??
Not sure if the activity above was an instance where it was supposed to
go to 0742242442 but the DID being passed to iaxmodem wasn't 0742242442:
Dial(SIP/5060-0a2f7308, IAX2/4111/0282086500,20,d) in new stack
In this case $CALLID4 is going to be 0282086500. Double check your
extensions.[conf|ael] and make sure the DID that was called is being
passed in your dial command, often like this: Dial(IAX2/4111/${EXTEN})
-Dave
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Re: [asterisk-users] IAXmodem in dialplan
On 06/07/2010 01:27 PM, Michelle Dupuis wrote: I'm succesfully using IAXmodem for faxing (using hylafax) with Asterisk. I would like a little more control for outbound calls using IAXmodem, but I'm not sure how to do it. It looks like dialing out over IAXmodem bypasses the dialplan altogether...can anyone confirm this? MD No, it does not bypass the dialplan. Asterisk treats is just like any other IAX endpoint. You need to specify the context in iax.conf for the entry IAXmodem is registering against. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] This is a test, hijack this
On 03/24/2010 03:56 PM, Miguel Molina wrote: Gergo Csibra escribió: Hello Asterisk, This is only a test, because I can't start new thread in this list... If you can send an email, you can start a new thread on this list. What's the point of all this? He was probably having the same problem I've had where I can reply to exiting threads fine but any time I send a fresh email to start a new thread it never goes through. Murphy's law being what it is this email that he suspected would never go through... did. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time/date over POTS?
Jeff LaCoursiere wrote: I had a customer ask me about time/date information being sent to his analog (attached to a Linksys SPA2102) answering machine. I didn't know that POTS could carry this information. Is this something Asterisk could send over SIP? Cheers, j Time and date info on a POTS line is part of the caller ID stream. It is up to the analog endpoint sending the caller ID stream to know the current time to send. Anything that works with SIP should also have NTP capabilities and should be getting its time using that. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time/date over POTS?
Jeff LaCoursiere wrote: On Thu, 4 Mar 2010, Dave Fullerton wrote: Jeff LaCoursiere wrote: I had a customer ask me about time/date information being sent to his analog (attached to a Linksys SPA2102) answering machine. I didn't know that POTS could carry this information. Is this something Asterisk could send over SIP? Cheers, j Time and date info on a POTS line is part of the caller ID stream. It is up to the analog endpoint sending the caller ID stream to know the current time to send. Anything that works with SIP should also have NTP capabilities and should be getting its time using that. -Dave Aha. Sadly I know that the incoming calls from our PSTN provider (over RBS T1) do NOT carry caller ID, so what we are passing on via SIP to the Linksys box must also be missing the time info. Is there any way to add that to the outgoing call to the Linksys box? Cheers, j The time and date in your case is being generated (or should be) by the sipura, not whatever is sending the call to the sipura. Time and date information is not included in SIP caller ID (to my knowledge). It's up to the SIP endpoint to know what time it is. Check the NTP settings on the sipura to make sure it is syncing its time with an internet time server. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Server suggestion
Sascha Ferley wrote: Hi, I am in need of ordering a new server here for our asterisk solution. Since the corporate standard is Dell we need to stick to a dell server. We used to deploy 2900III without any issues, however now they are almost not available any more and are looking at a new solution. Has anyone tried any of the new Dell R (series) servers with Asterisk, utilizing Digium PRI cards? The biggest issue I can see is that in the future we may want to get a transcoder card, however none of the new servers have a standard PCI slot available any more as with the new Nathelem chips having gotten rid of the basic bridge I guess. We're using an R200 with a TE220B dual T1 card for 8 months now without issue. As for the PCI problem, Digium makes a PCI Express transcoder card (TCE400B). As long as the server you buy has a built-in disk controller you should have two open PCI Express slots to play with. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
That's a bit misleading. Yes calls that travel over a PRI will be using
ulaw, but only over the PRI leg of the call. The SIP leg can still be
using G.729 with asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for the
peers you are working on but I have a guess as to what is going on. In
one of your previous messages you state: I moved G.729 to the top of
that list (just below disallow) I'm guessing your list looks something
like this:
disallow=all
allow=g729
allow=ulaw
allow={maybe something else}
This will be fine for all the phones in the office but the remote phones
need to ONLY have disallow=all and allow=g729 in their entries in
sip.conf as Jeff's reply stated. By having the allow=ulaw entry in there
you are giving asterisk permission to allow any call that is already in
the ulaw format (calls from the PRI) to remain in that format when
contacting your remote phones. If you're still stick post your sip.conf
(with the passwords removed) and we can help you out.
-Dave
Danny Nicholas wrote:
IMO you can only use the G.729 on a SIP call. If the call falls onto the
PRI framework, ulaw will be forced.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke...
What I'm hoping for is that all of the connections between my phones (or
at least a particular group of them) and my Asterisk server will use
G.729. Currently it seems like it usually is, but not always, and I
haven't figured out the pattern.
All of our calls fall into two categories:
Internal calls - one extension to another within our single Asterisk
server org.
External calls - To/From one of our extensions out thru the PRI line to
our carrier (Hawaiian Tel) to phone numbers out in the world.
For some reason it appears that inbound calls from out in the world are
going to our phones using ULAW, but outbound calls to the world are
using G.729.
That's progress but...how can I get my Asterisk server to use G.729 to
pass those incoming calls to my phones?
Best wishes and aloha,
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Tuesday, December 15, 2009 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
O.K., interestingly enough when I call our extensions from my mobile
phone it still seems to be using ULAW, but when they dial out it
seems
to be using G.729 now.
Is there something in Dahdi that I need to configure so that inbound
calls (from the PRI on a Digium TE205) use G.729 to get to the
phones
too?
A Dahdi channel over a PRI will always be ulaw - that is the encoding
on the
PRI (at least in the US). If your phones are using G.729 then
transcoding will
be taking place within asterisk for the bridge between the channels.
My guess is you are looking at the PRI channel. There should be
another
channel for the phone. That should always be G.729 now.
Cheers,
j
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of j...@jeff.net
Sent: Tuesday, December 15, 2009 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
Ahhh...yes, I think that may have been it. I moved G.729 to the
top
of that list (just below disallow) and now I have a restart when
convenient pending. Is that sufficient or do I have to actually
reboot the server for the change to take effect?
Just do a sip reload at the asterisk CLI prompt and you will be
good
to go. It
won't cutoff any calls in progress. Then reboot your phone.
Cheers,
j
___
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Re: [asterisk-users] Can't get G.729 to work...
I don't know how FreePBX works, but with vanilla Asterisk you would do
something like this with your sip.conf:
[general]
disallow=all
allow=ulaw
allow=g729
[localA]
callerid=Local phone A 100
username=localA
secret=blahblah1
[localB]
callerid=Local phone B 101
username=localB
secret=blah1blah
[remoteA]
callerid=Remote phone A 102
disallow=all
allow=g729
username=remoteA
secret=123456
[remoteB]
callerid=Remote phone B 103
disallow=all
allow=g729
username=remoteB
secret=654321
You can do this using templates as well, but this will make it easier to
understand. See the disallow/allow lines on the remote peers? Those
override the settings in the general portion of your sip.conf. With
these settings the local phones will use ulaw by default and allow g729
when needed.
This will do what you want for the most part. Local phones will use ulaw
for all calls between themselves and calls in and out of the PRI. Calls
from a remote phone to a local phone will use g.729 end to end. Calls
from a local phone to a remote phone will use ulaw between the local
phone and asterisk and g.729 between asterisk and the remote phone (this
is a limitation of asterisk's codec negotiation). Calls from remote
phones will use g.729 all the time.
I'm sure there is a way to do this through the freepbx gui, but like I
said, I have no experience with freepbx.
-Dave
Ben Schorr wrote:
O.K., I think I'm catching on. I only have a single SIP.CONF file that
ALL of the extensions are using so I'm gathering that I need to set up a
separate SIP.CONF file (or perhaps just an included file) for the 8
users at the remote office which ONLY Allows the G.729.
So now I'm figuring out how to do that.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
That's a bit misleading. Yes calls that travel over a PRI will be
using ulaw, but
only over the PRI leg of the call. The SIP leg can still be using
G.729 with
asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for the
peers
you are working on but I have a guess as to what is going on. In one
of your
previous messages you state: I moved G.729 to the top of that list
(just
below disallow) I'm guessing your list looks something like this:
disallow=all
allow=g729
allow=ulaw
allow={maybe something else}
This will be fine for all the phones in the office but the remote
phones need
to ONLY have disallow=all and allow=g729 in their entries in sip.conf
as Jeff's
reply stated. By having the allow=ulaw entry in there you are giving
asterisk
permission to allow any call that is already in the ulaw format (calls
from the
PRI) to remain in that format when contacting your remote phones. If
you're
still stick post your sip.conf (with the passwords removed) and we can
help
you out.
-Dave
Danny Nicholas wrote:
IMO you can only use the G.729 on a SIP call. If the call falls
onto the
PRI framework, ulaw will be forced.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke...
What I'm hoping for is that all of the connections between my phones
(or
at least a particular group of them) and my Asterisk server will use
G.729. Currently it seems like it usually is, but not always, and I
haven't figured out the pattern.
All of our calls fall into two categories:
Internal calls - one extension to another within our single Asterisk
server org.
External calls - To/From one of our extensions out thru the PRI line
to
our carrier (Hawaiian Tel) to phone numbers out in the world.
For some reason it appears that inbound calls from out in the world
are
going to our phones using ULAW, but outbound calls to the world are
using G.729.
That's progress but...how can I get my Asterisk server to use G.729
to
pass those incoming calls to my phones?
Best wishes and aloha,
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Tuesday, December 15, 2009 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Jon Moore wrote: On Tue, Nov 10, 2009 at 11:43 AM, Doug Lytle supp...@drdos.info wrote: Jon Moore wrote: I changed the line in extensions.conf to be SET(CALLERID(ALL)=Corner Homecare2703653903) To match what it appears I'm getting from ATT, only the 10 digit number. We've got ATT out of the Detroit area, you can't set callerid name, only number. So, try: exten = _91NXXNXX,1,Set(CALLERID(number)=12703653903) I noticed I had the number wrapped in angle brackets here, and removed those as well. Still having the issue though. Thanks for the pointer. Did you have to provide ATT with a list of numbers you would be setting, or does it Just Work? -jonathan I have an ATT PRI out of Holland and it just works. I actually have it pass the internal extension number as outbound called ID (which is 4 digits) when anyone calls my cell phone so I know how to answer and it works fine. This is what I'm using: set(CALLERID(num)=1234567890) Note num and not number I don't know if that was a change from 1.4 to 1.6 or if Doug mistyped it. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Libpri-1.4.10.2 Released
To my knowledge DAHDI does not use libpri, only asterisk. In my experience you can upgrade libpri and restart asterisk, just like you did, to make the upgrade take effect. As to what the proper thing to do is, it's probably better to recompile asterisk after upgrading libpri. -Dave Karl Fife wrote: Question about the proper way to update LibPRI: 'Bouncing' asterisk after an installing the new LibPRI version does indeed reflect the update: asterisk*CLI pri show version libpri version: 1.4.10.2 BUT some friendly chaps in the IRC channel have suggested that Asterisk Dahdi need to be recompiled as well. Any truth to this? Thanks -Karl On Tue, Oct 20, 2009 at 3:50 PM, Asterisk Development Team asteriskt...@digium.com wrote: The Asterisk Development team is pleased to announce the release of libpri 1.4.10.2, which is available for immediate download at: http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4.10.2.tar.gz This release resolves various issues found in libpri 1.4.10.1 and earlier versions related to scheduler events not being deleted and new ones being created on top of them. This can cause the scheduler to be overfilled, as well as other Q.921 related badness because of runaway scheduled events. Note, this can only happen when Q.931 messages are attempted to be sent during a D-Channel state transient (D-Channel goes down and back up). For a full list of changes in this release, please see the ChangeLog: http://svn.asterisk.org/svn/libpri/tags/1.4.10.2/ChangeLog Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing callerid on when using a PRI
Jason Parker wrote: Doug Lytle wrote: Dave Fullerton wrote: Note num and not number I don't know if that was a change from 1.4 to 1.6 or if Doug mistyped it. Not a mistype. I've been using number all along, but looking at the docs shows that I've been incorrect. It must concatenate the number down to num. Looks like I've got a little modifying to do this evening: core show function CALLERID livonia*CLI -= Info about function 'CALLERID' =- [Syntax] CALLERID(datatype[,optional-CID]) [Synopsis] Gets or sets Caller*ID data on the channel. [Description] Gets or sets Caller*ID data on the channel. The allowable datatypes are all, name, *num*, ANI, DNID, RDNIS. Uses channel callerid by default or optional callerid, if specified. Doug The documentation is correct, but the way the check really works, is that it reads the first 3 chars and matches it to num. This means that num, number, and numnumnumIloveapplesauce would all technically match. lol. Love it. I want to use that in my dialplan just to make my successor go WTF? -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Software Data Modem
I didn't read every post in that thread, but I don't think that's what he's asking about. I think what he wants (and I would like too) is something like iaxmodem that instead of connecting to hylafax you connect to pppd or minicom or the like. I'd love to be able to provide one or two channels of dialup access on my PRI to remote users. -Dave Danny Nicholas wrote: Start with this thread http://www.velocityreviews.com/forums/t233590-asterisk-with-modems-instead-o f-phonecards.html _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cherif Sent: Tuesday, November 03, 2009 7:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Software Data Modem Hello everybody I am trying to connect my asterisk to a payment equipment trough PSTN. I have a TDM400P card with an fxs module an the equipment use modem to send data! I was thinking to implement a software data modem in asterisk, but I found out that there is just faxmodem for asterisk, Is anyone here know something about software data modem working with asterisk to help out? Thanks, Regards Mosleh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to organize TFTP root directory ?
Olivier wrote: Hi, Most (if not all) IP phones support provisioning through DHCP/TFTP. The trouble is some phones seem to require to store their config files in TFTP root directory. This makes this TFTP root directory a bit messy. What are the best practices or tricks to manage this TFTP root directory ? I was thinking of either : 1. building a dedicated source TFTP tree in which files are cleanly organized (vendor/models:...) which would be synchronized (one way ? two ways ?) with the official TFTP tree (that would be then, collapsed to a single directory) 2. tune DHCP/TFTP server config so that each phone would retrieve its config files from a vendor-dedicated subdirectory. I don't have a clue about solution 2. Is it even possible ? Solution doesn't look very encouraging as it might be difficult to keep trees in sync. #2 might be possible, but there's a lot of depends on factors. The ISC dhcpd often packaged in linux distributions has the ability to specify different dhcp options to different pools of addresses. You can then assign clients to pools based on a substring match of their mac address. This then requires that the client (phone) will use the URL specified in dhcp option 66. With all this put together you can assign each brand of phone to its own pool/options where the options point it to a URL containing the firmware for that brand of phone. I do this with my polycom phones and it works well. Don't know if it works with other brands of phones. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT wanted old Sipura firmware 2.0.13
Joseph wrote: Does anybody know here I can find old Sipura firmware 2.0.13 for SPA-3000 I have Cisco 3.1.20 but it is not working as it suppose to. http://www.totek.ca/index.php?option=com_contenttask=viewid=151Itemid=39 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS/DSCP for IAX?
Michelle Dupuis wrote: I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks Yes the tos setting is the right place and EF is an acceptable value. EF is the differentiated services code point (or dscp) for expedited forwarding. The sample sip.conf defaults tos_audio to EF as well. The iax.conf wiki page only shows the old type of service values which are considered deprecated. Look at this page for more info on diffserv: http://www.voip-info.org/wiki/view/DiffServ As for what to use, well, that depends on whether your upstream provider even honors what you set. They may use the old type of service values, they may use dscp or they may ignore what you put there entirely. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS/DSCP for IAX?
Yea, kinda, sorta. The DSCP is six bits, which occupy six of the 8 bits in what is/was the type of service byte in an IP packet. Three of the 6 DSCP bits reside over the old precedence field and three reside over the old low delay, high throughput and high reliability fields (those three often referred to as TOS). The DSCP code points are designed to be backwards compatible with the PRECEDENCE portion of the old tos. The low delay, high throughput and high reliability bits have been redefined and no longer are backwards compatible. When doing my research I found some web sites displayed the tos byte in different bit-orders (cisco with precedence first, wikipedia with precedence last). It was confusing as heck. I also have some old equipment that does not understand DSCP/Diffserv. What I ended up doing was making asterisk and phones use the dscp code points and my old router software queue packets based on what it sees in the precedence field. Works like a charm. Good luck. -Dave Michelle Dupuis wrote: That link is great thanks. From what I read elsewhere, ToS is just the first 3 bits which should be honored by DSCP (first 5 bits)- even old equip should be DSCP compatible...or I need to do more reading :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, October 01, 2009 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] QOS/DSCP for IAX? Michelle Dupuis wrote: I actually see the TOS setting in iax.conf, but the default (commented out) is EF - which doesn't even match a valid bit combination according to voip-info wiki If this is the right place, what TOS value are people using succesfully over an ADSL connection? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 01, 2009 2:27 PM To: Asterisk Users List Subject: [asterisk-users] QOS/DSCP for IAX? Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks Yes the tos setting is the right place and EF is an acceptable value. EF is the differentiated services code point (or dscp) for expedited forwarding. The sample sip.conf defaults tos_audio to EF as well. The iax.conf wiki page only shows the old type of service values which are considered deprecated. Look at this page for more info on diffserv: http://www.voip-info.org/wiki/view/DiffServ As for what to use, well, that depends on whether your upstream provider even honors what you set. They may use the old type of service values, they may use dscp or they may ignore what you put there entirely. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disable dtmf on SIP peer
Giedrius Augys wrote: Hello, I have one problem and I need to disable dtmf (disable rfc2833, info and inband) on one (other peers must support dtmf) SIP peer . Is it possible? Workaround would be use g729 codec with dtmfmode=inband. Maybe there is better solution? Thanks for help. Assuming you have control over the peer, simply set the peer to use rfc2833 and have asterisk listen for info (or other way around) on that peer. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
Mike wrote: Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get Push message cannot be displayed back from the Polycom phone, and all I am sending is the Polycom example : PolycomIPPhone Data priority=”critical” h1 Fire Drill at 2pm /h1 Please exit and congregate at your appropriate location outside /Data /PolycomIPPhone Using curl to send it to the phone (192.168.1.54/push) on the LAN as a first test. (all urlencoded, yes) Did anyone ever succeed in doing this here? I'd appreciate any tips. Mike I've never done it (or heard of it until now), it looks pretty cool. Is the apps.push.messageType field set in sip.cfg? Did you set the apps.push.username and apps.push.password fields and is curl sending that username/password to the phone? Just stabs in the dark. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
In case it's important to you, microbrowser support was added to the 501 and 430 back in SIP 2.1.0. Though how you could use a microbrowser on a 430 for much I don't know. -Dave Danny Nicholas wrote: This is also a stab-in-the-dark as my 501 doesn't have a microbrowser; Have you tried communicating with the phone via telnet to debug the problem? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Thursday, September 24, 2009 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom push application for microbrowser Mike wrote: Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get Push message cannot be displayed back from the Polycom phone, and all I am sending is the Polycom example : PolycomIPPhone Data priority=critical h1 Fire Drill at 2pm /h1 Please exit and congregate at your appropriate location outside /Data /PolycomIPPhone Using curl to send it to the phone (192.168.1.54/push) on the LAN as a first test. (all urlencoded, yes) Did anyone ever succeed in doing this here? I'd appreciate any tips. Mike I've never done it (or heard of it until now), it looks pretty cool. Is the apps.push.messageType field set in sip.cfg? Did you set the apps.push.username and apps.push.password fields and is curl sending that username/password to the phone? Just stabs in the dark. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo
Jason Baker wrote: Echo Cancellation: 128 taps unless TDM bridged, currently ON The currently ON is telling you that the echo canceller is active. You could try changing echotraining to no in chan_dahdi.conf as well. What were you running before you upgraded? So, Asterisk doesn't start echo canceling a line until it is in use? I thought that might be the case. I was running Zaptel before this, not sure what version. I upgrade to Dahdi. The echo was present in Zaptel, but not as bad. Does anyone have any experience with hardware echo cancel modules? Are they better/worse than software? What would be the best solution to remove echo? No, asterisk does not start echo canceling on a channel until the channel is brought up. I have two sites with hardware echo cancellers. One Digium on a TE220B and one on a Sangoma A200. I use OLSEC (software) on my home system. From my experience, I would say the hardware solutions work the best, though OSLEC is very good. I'm no echo expert, but if you are hearing your own voice echoed back to you on calls the first thing I would check is your txgain settings. There's plenty of info on voip-info.org to help you with that. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo
Jason Baker wrote: I recently upgraded my Asterisk system to Dahdi and now I have an echo problem. I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium TE121B PCI express card with a HARDWARE echo cancellation module. All this is housed on a CentOS 5.5 box, 2.6.18 Kernel. My incoming phone service is an ATT PRI (24 channel T1). My configs: chan_dahdi.conf* [channels] ; configuration for T1 card as PRI language = en group = 1 echocancel = yes echotraining = yes signalling = pri_cpe switchtype = 4ess usecallerid = yes context = incoming channel = 1-23 ***/etc/dahdi/system.conf* loadzone=us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 When I run dahdi_cfg -vvv I get the following: DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.1 Echo Canceller(s): MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: none) (Slaves: 01) snip Channel 23: Clear channel (Default) (Echo Canceler: none) (Slaves: 23) Channel 24: D-channel (Default) (Echo Canceler: none) (Slaves: 24) 24 channels to configure. Setting echocan for channel 1 to none snip Setting echocan for channel 24 to none It is showing MG2 as the echo canceller, even though I don't have an echo canceller specified. Is that the harwdare module? Do I even need to specify an echo canceller in the configs if I have a hardware echo module? MG2 is a software canceller. I don't think that line means that MG2 is being used on all your channels. If you look at the Channel map it says (Echo Canceler: none). If it had been set to MG2 you would see MG2 instead of none. You do not need to specify an echo canceller in system.conf when you have a hardware canceller. One thing I would check is to make sure asterisk is activating the echo canceller when a call is in progress. To do this execute core show channels at the asterisk command line (make sure someone on the system has placed a call on the PRI). Look for a DAHDI/#-x line. Then execute dahdi show channel # where # is the channel number. You'll get a screen full of output. Look for a line that looks like this (it will be near the end): Echo Cancellation: 128 taps unless TDM bridged, currently ON The currently ON is telling you that the echo canceller is active. You could try changing echotraining to no in chan_dahdi.conf as well. What were you running before you upgraded? -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Connected to definity errors
C F wrote: We have setup asterisk to handle our calls before between telco and an Avaya definity. The PRI keeps locking up every so often. In addition I keep getting this error when trying to call the avaya: -- Channel 0/2, span 1 got hangup request, cause 102 -- Hungup 'Zap/2-1' When that error happens I get a fast busy (congestion) tone. Any one can point me in the right direction? TIA I can't offer any help as to what is causing it, but according to this: http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php Cause No. 102 - recovery on timer expiry. This cause indicates that a procedure has been initiated by the expiration of a timer in association with error handling procedures. What it means: This is seen in situations where ACO (Alternate Call Offering) is being used. With this type of call pre-emption, the Telco switch operates a timer. For example, when an analog call is placed to a Netopia router that has two B Data Channels in place, the router relinquishes the second channel, but if it doesn't happen in the time allotted by the switch programming, the call will not ring through and will be discarded by the switch. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 single span TDM cards in asterisk
C F wrote: I need to add a second T1 to an asterisk system. However the first card is in a PCI-e slot, and the only available slot is a PCI card. Could that work? TIA Technically, you should be able to run two cards in two different type slots with no problem. You will double the number of interrupts your system has to handle. Also if you have to route modem/fax calls between the two spans you may run into trouble since the timing could be different. In that case you'll want a timing cable (if the cards support it) or one dual-span card. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
Tzafrir Cohen wrote: On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote: Here's my $0.02. If you don't want an echo canceller, specify echocanceller=none,x-y and have dahdi_cfg print a warning (at any verbosity level) when an echo canceller is not specified for a channel. Personally, I would also like to see an option that says Use the hardware canceller, like echocanceller=hw,x-y. This would have the added benefit of being able to display an error/warning when the hardware canceller is specified but no hw canceller is present. It goes against my grain to not specify a canceller to mean use a harware one if it happens to exist. Though this means you have to explicitly configure hardware echo cancellers to work, which is not as before. This leaves even more room for error. It is true that this method would require more configuration work and that it would probably throw people off who were used to the old method. However, I don't agree that it leaves more room for error. The current system, IMHO, has a certain amount of ambiguity to it. If I inherit a production system from someone, I can't tell for sure what the echo canceller setup is just by looking at system.conf. I have to look at system.conf and then know if hardware echo can is present. Aside from opening the case or looking at dmesg output, I'm not even sure how to see if a hardware echocan is present or not. The post that started this thread is another example of that ambiguity. Not defining an echo canceller to mean don't use one, or use a hardware one if there is one I think leaves room for confusion and error. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
Jeff LaCoursiere wrote: On Wed, 19 Aug 2009, Dave Fullerton wrote: Tzafrir Cohen wrote: On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote: Here's my $0.02. If you don't want an echo canceller, specify echocanceller=none,x-y and have dahdi_cfg print a warning (at any verbosity level) when an echo canceller is not specified for a channel. Personally, I would also like to see an option that says Use the hardware canceller, like echocanceller=hw,x-y. This would have the added benefit of being able to display an error/warning when the hardware canceller is specified but no hw canceller is present. It goes against my grain to not specify a canceller to mean use a harware one if it happens to exist. Though this means you have to explicitly configure hardware echo cancellers to work, which is not as before. This leaves even more room for error. It is true that this method would require more configuration work and that it would probably throw people off who were used to the old method. However, I don't agree that it leaves more room for error. The current system, IMHO, has a certain amount of ambiguity to it. If I inherit a production system from someone, I can't tell for sure what the echo canceller setup is just by looking at system.conf. I have to look at system.conf and then know if hardware echo can is present. Aside from opening the case or looking at dmesg output, I'm not even sure how to see if a hardware echocan is present or not. The post that started this thread is another example of that ambiguity. Not defining an echo canceller to mean don't use one, or use a hardware one if there is one I think leaves room for confusion and error. -Dave I feel like I must be missing something here. In 1.4, to my knowledge, if hardware echo cancellation was present, it would be used automatically. Further, software echo was enabled by default. If hardware was available the software would turn itself off automatically. What was wrong with this setup? There was no ambiguity, and there was no confusion. Have I assumed the above in error all this time? So in 1.6 the hardware echo is on if available, and its only that you must enable software cancellation if you want it by adding the appropriate module. Is that right? It seems then that we would be back to the 1.4 situation if asterisk shipped with one of the SEC modules enabled by default, and you could change it or turn it off if you wanted. Kevin seemed to confirm that this was the plan. Sounds good to me. Sort of, except it's not a difference between 1.4 and 1.6, it's a difference between Zaptel and DAHDI (which also works in asterisk 1.4). In Zaptel you compiled in a software echo canceller and that was used if a hardware canceller was not present (you didn't have to specify). In DAHDI, you must explicitly specify what software echo canceller you want to use for each channel in system.conf. If you do not specify, then you either do not get an echo canceller, or you automatically use the hardware canceller-if it is present. My suggestion is that you always have to explicitly state what echo canceler you wish to use for each channel, whether it be software, hardware or none at all. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
Kevin P. Fleming wrote: Dave Fullerton wrote: It is true that this method would require more configuration work and that it would probably throw people off who were used to the old method. However, I don't agree that it leaves more room for error. The current system, IMHO, has a certain amount of ambiguity to it. If I inherit a production system from someone, I can't tell for sure what the echo canceller setup is just by looking at system.conf. I have to look at system.conf and then know if hardware echo can is present. Aside from opening the case or looking at dmesg output, I'm not even sure how to see if a hardware echocan is present or not. The dahdi_scan tool will tell you whether hardware echocans are present or not, among other methods. I tried that, but I didn't see anything that specified whether the echo canceller was present. Here's the output, can you tell me what I should be looking for? r...@srv210394:~# dahdi_scan [1] active=yes alarms=OK description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=24 irq=16 type=digital-T1 syncsrc=2 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [2] active=yes alarms=OK description=T2XXP (PCI) Card 0 Span 2 name=TE2/0/2 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=25 totchans=24 irq=16 type=digital-T1 syncsrc=2 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF Thanks, Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
Kevin P. Fleming wrote: Dave Fullerton wrote: The dahdi_scan tool will tell you whether hardware echocans are present or not, among other methods. I tried that, but I didn't see anything that specified whether the echo canceller was present. Here's the output, can you tell me what I should be looking for? r...@srv210394:~# dahdi_scan [1] active=yes alarms=OK description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) It would show up in the 'devicetype' line, which should end with 'with VPM400M', 'with VPMOCT064' or 'with VPMOCT128' in the case of a dual/quad span card, depending on which module is attached. I guess I just found a bug then, because the card above is a TE220B. Here's a portion of the dmesg output: wct4xxp :02:08.0: PCI INT A - GSI 16 (level, low) - IRQ 16 Found TE2XXP at base address dfcfff80, remapped to f8872f80 TE2XXP version c01a016c, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x35b55400 Reg 1: 0x35b55000 Reg 2: 0x Reg 3: 0x Reg 4: 0xff01 Reg 5: 0x Reg 6: 0xc01a016c Reg 7: 0x1000 Reg 8: 0x Reg 9: 0x00ff00ff Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! dahdi: Registered tone zone 0 (United States / North America) About to enter startup! TE2XXP: Span 1 configured for ESF/B8ZS wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present timing source auto card 0! firmware: requesting dahdi-fw-oct6114-064.bin VPM450: echo cancellation for 64 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
Kevin P. Fleming wrote: Jeff LaCoursiere wrote: On Tue, 18 Aug 2009, Kevin P. Fleming wrote: [snip] Note: It is *mandatory* to configure an echo canceler for the system's channels using dahdi_cfg unless the interface cards in use have echo canceler modules available and enabled. There is *no* default software echo canceler with DAHDI. Why is this by the way? Is there some advantage to NOT having one of these modules loaded by default? Well, when we made them modular so that people could pick and choose at run-time instead of compile-time, it seemed like forcing a default on everyone was the wrong thing to do... especially for people who don't need them at all because they have hardware echocancelers. In hindsight, this has probably been the biggest issue with people upgrading from Zaptel to DAHDI, and we should have just had some sort of default. We've had some discussions about making dahdi_cfg supply a default echo canceler for all channels that don't have one specified, but then that of course will require the ability to tell it no, I don't want one. Here's my $0.02. If you don't want an echo canceller, specify echocanceller=none,x-y and have dahdi_cfg print a warning (at any verbosity level) when an echo canceller is not specified for a channel. Personally, I would also like to see an option that says Use the hardware canceller, like echocanceller=hw,x-y. This would have the added benefit of being able to display an error/warning when the hardware canceller is specified but no hw canceller is present. It goes against my grain to not specify a canceller to mean use a harware one if it happens to exist. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating an IAX/SIP-to-ISDN PRI gateway
Shashi Dookhee wrote: Hi all, I'd like to setup a really lean Asterisk installation that essentially has a full ISDN PRI (ATT, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan) on a Digium TE207P adapter that all it does is convert the ISDN channels to SIP/IAX channels. Then I would add this Asterisk 'gateway' as a provider on one (or many) Asterisk systems on the back. With such a config I don't need anything like Voicemails, mailboxes, etc... All I want it to do is accept calls and 'passthru' the caller ID, and when it receives a call, send it to the appropriate Asterisk server based on Called ID (and, of course, passthru that 'Called ID' too). Any help is appreciated - the Asterisk config files are overwhelming and we need to get this done pretty quickly! Thanks in advance for your help! Not exactly sure what information you're asking for, but here's a starting point. You'll need the latest DAHDI, libpri and asterisk (I'd grab 1.4.26.1 myself). Compile and install each in turn. As for configuration files, you should only need to worry about the following: /etc/dahdi/system.conf (this will get you started) # define spans span=1,0,0,esf,b8zs bchan=1-23 dchan=24 # Global Options loadzone=us defaultzone=us in /etc/asterisk: You can either make samples to install all sample files or you can copy the sample files from /usr/src/asterisk-1.4.26.1/configs/ (or wherever you extracted asterisk from). asterisk.conf (use the sample and tweak if needed) chan_dahdi.conf should look something like this: [channels] context=inbound-pri switchtype=national pridialplan=unknown resetinterval=never signalling=pri_cpe group=1 channel=1-24 extensions.conf (you'll find plenty of examples online) iax.conf (start with sample config and tweak to your liking) sip.conf (again, start with sample and tweak) logger.conf (sample will work) modules.conf (start with sample) indications.conf (use the sample) Good Luck -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI failover to SIP trunk
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
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Re: [asterisk-users] PRI failover to SIP trunk
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
It is my understanding (through trial and error, reading, etc) that any
Dial command that does not result in an answered state will continue in
the dialplan after a timeout (if specified) or some sort of progress is
received. If the called channel results in an answer then dialplan
processing stops as soon as one party hangs up (unless the g option is
specified).
This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
as the dial is complete so you won't be able to use this trick under
normal circumstances.
-Dave
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Re: [asterisk-users] PRI failover to SIP trunk
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
While this will work, be aware that there are circumstances where you
may end up calling the number twice, once through each provider. One
example is if the number you dial is busy, that progress will be passed
via the PRI to asterisk and the dialplan will continue to the next
priority. In this case, dialing the number again through the SIP
provider. To avoid this you will need to use some dialplan logic and
check the result of the DIALSTATUS variable. See this page for examples:
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Dave
Good point.
I was unaware that busy back from a TDM circuit would progress in the
dialplan rather than going to the h exten.
What other cases are there like that?
It is my understanding (through trial and error, reading, etc) that any
Dial command that does not result in an answered state will continue in
the dialplan after a timeout (if specified) or some sort of progress is
received. If the called channel results in an answer then dialplan
processing stops as soon as one party hangs up (unless the g option is
specified).
This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
as the dial is complete so you won't be able to use this trick under
normal circumstances.
-Dave
True I guess except that if the call fails as the OP posted, because the PRI
is down, it should work then right?
I believe so, I haven't tried it. I imagine DIALSTATUS would be either
CHANUNAVAIL or CONGESTION.
Another thing. For outbound calls, I do not have a timeout. So the user
hangs up when they are ready, or when the other side hangs up or gets
congestion, which amounts to the h exten, or am I not correct.
I can't answer to the use of the h exten, I've never used it.
Why have a timeout on outbound dialing (unless you are a dialer app?) It is
not like voicemail where you want it to ring for so many seconds and then
roll to VM.
You usually wouldn't use a timeout for outbound PSTN calls. I only
mentioned it to try to be as complete as possible.
-Dave
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Re: [asterisk-users] g.722 + loudness
Kevin P. Fleming wrote: Hose wrote: I have a feeling that the issue is between transcoding of ulaw to g.722 and it's too loud during the transcoding - anyway to adjust the levels? There was a flaw in Asterisk's G.722 transcoder module that was fixed recently (on May 15, 2009), so any release made after that date should solve your problem. Upgrading to 1.6.0.10 should give you the fix (and the fix should be noted in the ChangeLog for 1.6.0.10 as well). It is not in the 1.6.0.10 Changelog nor the 1.6.1.1 Changelog. It is, however, in the 1.6.0.11-rc1 Changelog and the 1.6.2.0-beta3 Changelog. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ?
Olivier wrote: Hi, I'm wondering how I could notify to a dumb analog phone that a voicemail message is waiting. My goal would be to change the tone that is heard just before user starts to dial. Any idea on that ? Yea, it's called stutter dial tone. For DAHDI channels just specify the mailbox in chan_dahdi.conf. If it's connected to an ATA then specify the mailbox on the peer in sip.conf/iax.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem configuring TDM400
jonas kellens wrote: On Fri, 2009-07-03 at 11:58 +0100, Mike wrote: tempest:~# lspci 00:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I don't think this is you TDM-card... This is mine : 04:05.0 Ethernet controller: Digium, Inc. TDM400P (rev 11) Subsystem: Digium, Inc. TDM400P Flags: bus master, medium devsel, latency 32, IRQ 90 I/O ports at d100 [size=256] Memory at ff74 (32-bit, non-prefetchable) [size=1K] Expansion ROM at 8000 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 I don't think your XEN VM can see your TDM-card. You will need to ad a module to your XEN-kernel to be able to speak to your TDM pci-card. Don't know if this module exists... My TDM400P appears as his does: 00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem configuring TDM400
Mike wrote: Folks, I have a Xen Asterisk VM with a TDM400 card. When I try to run dahdi_cfg, I get: tempest:~# dahdi_cfg -vvv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0 Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) Channel 03: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 04) 3 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) The card appears to be detected: tempest:~# lspci 00:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I have the kernel modules loaded: tempest:~# lsmod Module Size Used by wctdm 35024 0 dahdi 185352 1 wctdm crc_ccitt 2848 1 dahdi autofs418500 0 ipv6 236612 10 ext3 106664 1 jbd43092 1 ext3 mbcache 8260 1 ext3 dm_mirror 16288 0 dm_log 9444 1 dm_mirror dm_snapshot15108 0 dm_mod 47304 3 dm_mirror,dm_log,dm_snapshot raid1 19200 0 md_mod 69180 1 raid1 thermal_sys11624 0 [ 1327.030178] dahdi: Telephony Interface Registered on major 196 [ 1327.030253] dahdi: Version: 2.2.0 I have Googled for this problem and found a lot of people reporting the issue but nobody really having much of an answer! I've seen the issue a few times. The strange thing is that I did have things working but then I rebotoed the box and it seems to have given up. I have a fairly straight forward DAHDI config file which has served me perfectly well in the past. tempest:~# cat /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Fri Jul 3 09:56:12 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Global data loadzone= uk defaultzone = uk fxoks=1 fxsks=3,4 I have tried only bringing up certain channels but that still fails. Does anyone have any idea what could be wrong? Mike. Did do all the device files show up in /dev/dahdi/ ? You should have something close to this: r...@jaguar:~# ls -l /dev/dahdi/ total 0 crw-rw 1 asterisk asterisk 196, 1 2009-07-05 09:32 1 crw-rw 1 asterisk asterisk 196, 2 2009-07-05 09:32 2 crw-rw 1 asterisk asterisk 196, 3 2009-07-05 09:32 3 crw-rw 1 asterisk asterisk 196, 4 2009-07-05 09:32 4 crw-rw 1 asterisk asterisk 196, 254 2009-07-05 09:32 channel crw-rw 1 asterisk asterisk 196, 0 2009-07-05 09:32 ctl crw-rw 1 asterisk asterisk 196, 255 2009-07-05 09:32 pseudo crw-rw 1 asterisk asterisk 196, 253 2009-07-05 09:32 timer -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimizing downtime during updates
- Original Message - From: Dave Fullerton dfullertaster...@shorelinecontainer.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 23, 2009 8:39 AM Subject: Re: [asterisk-users] Minimizing downtime during updates Karl Fife wrote: I was about to ask this question when I figured out the answer by combing through the makefile. I am posting this anyway because I think it's good to know, and I didn't find any threads that speak to it when I searched the list history. My Question was: When updating Asterisk, the sound tarballs for the selected codecs are not retreived until running make install. This adds unnecessarily to the downtime when updating versions because Asterisk has to be stopped while running make install. I wanted a simple way to pre-fetch these files to a local repository to speed up the actual install routine, instead of slowing it by the arbitrary duration of the fetch/download process which robs valuable NINES from uptime :-) I discovered that after running make, you can run 'make sounds' before shutting down the service. This cuts all of the download time from the install process minimizing service downtime to a fraction of what it would othewise be. You can also just grab and un-tar the sound files by hand from: http://downloads.asterisk.org/pub/telephony/sounds/ On a side note, why does the sounds directory not display in the directory listing when looking at http://downloads.digium.com/pub/telephony/ ? -Dave Karl Fife wrote: Dave Fullerton dfullertaster...@shorelinecontainer.com wrote: You can also just grab and un-tar the sound files by hand from: http://downloads.asterisk.org/pub/telephony/sounds/ Good point because the MOH files would need to be bulled down manually if you want to minimize downtime AND you choose to offer wideband MOH tracks to calling parties (and perhaps other native codecs). It is my observation that make sounds target does not fetch those MOH tracks (as I would have expected), rather they are only fetched during 'make install', (increasing downtime). Does anyone know if there is in fact a distinct target in the makefile that pulls these down, and if not, why they're not pulled down as a matter of course with make sounds if specified in makefile.makeopts. -Karl The trigger (I believe) is when you select the sound packages in menuselect. By default the GSM core sound files and WAV music on hold are included in the asterisk tar file. If you want to pre-download music files then wget them into the /usr/src/asterisk-1.x.x/sounds directory prior to running make. By skimming through the make file in that directory it looks like it tests for their existence prior to downloading them. Make sure you download the version appropriate to that versions of asterisk (you'll have to look in the sounds/Makefile at CORE_SOUNDS_VERSION and EXTRA_SOUNDS_VERSION) and not the -current tar balls. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimizing downtime during updates
Karl Fife wrote: I was about to ask this question when I figured out the answer by combing through the makefile. I am posting this anyway because I think it's good to know, and I didn't find any threads that speak to it when I searched the list history. My Question was: When updating Asterisk, the sound tarballs for the selected codecs are not retreived until running make install. This adds unnecessarily to the downtime when updating versions because Asterisk has to be stopped while running make install. I wanted a simple way to pre-fetch these files to a local repository to speed up the actual install routine, instead of slowing it by the arbitrary duration of the fetch/download process which robs valuable NINES from uptime :-) I discovered that after running make, you can run 'make sounds' before shutting down the service. This cuts all of the download time from the install process minimizing service downtime to a fraction of what it would othewise be. You can also just grab and un-tar the sound files by hand from: http://downloads.asterisk.org/pub/telephony/sounds/ On a side note, why does the sounds directory not display in the directory listing when looking at http://downloads.digium.com/pub/telephony/ ? -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP/SIP traffic prioritization and Linux issues
John A. Sullivan III wrote: Hello, all. I've stumbled across what seems to be a traffic prioritization issue in a Linux environment and wonder if anyone else has encountered or addressed this issue. We had planned to use expedited forwarding for our RTP and perhaps our SIP packets. Our plan was to set DSCP to 101110 (by the way, I think document http://www.voip-info.org/wiki/view/snom+360 is in error as I'm almost certain the expedited forwarding bits are 101110 and not 100010). However, we realized that when these passed through Linux based routers or firewalls using the default pfifo_fast packet scheduler, it would look at bits 3-7 for placement in band 0, 1, or 2. Using the standard expedited forwarding DSCP means pfifo_fast will see 1100 and place the packets in band 1 - the default band for all traffic. Thus, they will receive no prioritization. We are planning to thus change the DSCP to 101100 (b0 instead of b8 for Asterisk, 176 instead of 184 for our Snom phones) and map 101100 to 802.1p priority 7 on our switches. I am imagining this or is it a real issue when using Linux based firewalls and routers with default packet schedulers and expedited forwarding? Thanks - John You are correct, EF is 101110. I recently started using dscp on my network and ran into similar issues as you. I have cisco routers (not on smartnet) in my environment and some (v 12.x) understood dscp and some (=v 11.x) did not. For those that did not I had to match on the precedence bits instead and everything thus far is working like it is supposed to. As for linux, I couldn't find anything online that actually implemented diffserv-style traffic management. I ended up writing a script that would generate a set of queues and used the dscp to drop packets into the appropriate queues and another script to set the dscp for programs that could not on their own. It's still a bit of a work in process and I'm sure there are improvements to be made, but if you'd like to look at it I can send it to you off-list. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue
James A. Shigley wrote:
snip
The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for got
hangup, cause 50. What is cause 50?
snip
According to these sites:
http://www.quintum.com/support/xplatform/ivr_acct/webhelp/Disconnect_Cause_Codes.htm
http://www.cisco.com/en/US/docs/ios/11_0/debug/command/reference/disdn.html
Cause code 50 is:
Requested facility not subscribed - The remote equipment supports the
requested supplementary service by subscription only.
I don't know what that really means, sorry. It could be a setting with
your switchtype or pridialplan in chan_dahdi.conf. Or, it could be
something's not set up right on the telco side.
-Dave
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Re: [asterisk-users] Polycom registration errors
Jim Gottlieb wrote: I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from 'sip:6193644...@jtsd05' failed for '192.168.200.99' - Username/auth name mismatch Turning on SIP debug, it appears it's asterisk trying to register with the phone: Using latest REGISTER request as basis request Sending to 192.168.200.99 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.200.99:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.200.99;branch=z9hG4bKd732a3486F528693;received=192.168.200.99 From: 6193644850 sip:6193644...@jtsd05;tag=A1BB38FF-7161AAEA To: sip:6193644...@jtsd05;tag=as3d68239c Call-ID: 20f907fe-db323389-f4569...@192.168.200.99 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 But then, the From: and To: lines seem to both show it from hostname jtsd05, though there's also the line saying it's going to 192.168.200.99 (the phone). I've played with all sorts of settings in sip.conf, but the messages persist. Here's what I've got: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 disallow=all allow=ulaw dtmfmode=rfc2833 progressinband=no ;Polycom phones have trouble with the progressinband=never callerid=HFT Booth 0 (619) 364-4850 allowsubscribe=yes And some of the Polycom phone config: reg reg.1.displayName=HFT0 reg.1.address=6193644850 reg.1.label=4850 reg.1.type=private reg.1.lcs= reg.1.csta= reg.1.thirdPartyName= reg.1.auth.userId=hft0 reg.1.auth.password=mysecret reg.1.auth.optimizedInFailover= reg.1.musicOnHold.uri= reg.1.server.1.address=jtsd05 reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires= reg.1.server.1.expires.overlap= reg.1.server.1.register= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.server.1.lcs= reg.1.outboundProxy.address= Try changing reg.1.address to hft0. My hunch is asterisk is looking at the from of 6193644...@jtsd05 and going huh? I don't know a 6193644...@jtsd05. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26-rc1 Now Available
Asterisk Development Team wrote: The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc1 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release is primarily a fix for an issue (#14867, #14717) related to security fix AST-2009-001 where IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames. Additionally, an issue Excellent timing. I was just setting up an IAX account (incorrectly) in Zoiper and my console was being flooded by registration failure messages. As I was scratching my head going what the heck, this email magically pops in my inbox explaining it. Wish that worked with my bills... -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI fun and games
Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Wednesday, May 20, 2009 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI fun and games Danny Nicholas wrote: Hi Listers, I'm running 1.4.25-rc1 on opensuse 11.0 with dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2. Incoming calls work fine. Outgoing calls made directly (exten = s,1,Dial(DAHDI/G1) then number work fine. The problem I have is trying to let Asterisk make the call (exten = s,1,Dial(DAHDI/G1/5551212,,r). If I use m (moh) the music plays 5-8 seconds after the other end picks up. When using r, I get 2-3 rings after other end picks up. I've went through every flavor of dahdi-linux from 2.0.0 to 2.1.0-rc4 (which crashed me) with no joy. Any suggestions? Hardware is Dell Poweredge 1650/1550 and TDM410P/TDM400P. Any reason you're using the r/m option at all? Since this is an analog card I would leave the r/m off and just let asterisk use the in-band progress from the telco. -Dave Using r/m because DAHDI takes 10-15 seconds to get TELCO rings. My experience with analog channels has been that DAHDI will bridge audio immediately after dialing the last digit. The exception to that may be if you're trying to use callprogress=yes in chan_dahdi.conf. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI fun and games
Danny Nicholas wrote: Hi Listers, I'm running 1.4.25-rc1 on opensuse 11.0 with dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2. Incoming calls work fine. Outgoing calls made directly (exten = s,1,Dial(DAHDI/G1) then number work fine. The problem I have is trying to let Asterisk make the call (exten = s,1,Dial(DAHDI/G1/5551212,,r). If I use m (moh) the music plays 5-8 seconds after the other end picks up. When using r, I get 2-3 rings after other end picks up. I've went through every flavor of dahdi-linux from 2.0.0 to 2.1.0-rc4 (which crashed me) with no joy. Any suggestions? Hardware is Dell Poweredge 1650/1550 and TDM410P/TDM400P. Any reason you're using the r/m option at all? Since this is an analog card I would leave the r/m off and just let asterisk use the in-band progress from the telco. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels configuration with DAHDI
Daniel Bareiro wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Tzafrir. El miércoles 20 de mayo del 2009 a las 10:00:46 -0300, Tzafrir Cohen escribió: On Wed, May 20, 2009 at 07:03:15AM -0300, Daniel Bareiro wrote: Hint: you don't need to set 'signalling' for analog channels. Or just set it explicitly to auto. This is for Asterisk = 1.6.0 . Simply reduces the complication a bit... Thanks for the tip. I will remember it for when I use Asterisk 1.6 :-) I load the modules wctdm and dahdi. But when I execute in Asterisk CLI dahdi show channels, I get the following error message: No such command 'dahdi show channels' (type 'help dahdi show' for other possible commands) Try running: asterisk -r and in that prompt: module unload chan_dadhi.so module load chan_dadhi.so and tell us the output you got. # asterisk -r Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.4.24.1 currently running on alderamin (pid = 19777) Verbosity is at least 7 alderamin*CLI alderamin*CLI module unload chan_dadhi.so alderamin*CLI module load chan_dadhi.so [May 20 17:52:19] WARNING[10345]: loader.c:359 load_dynamic_module: Error loading module 'chan_dadhi.so': /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: No such file or directory [May 20 17:52:19] WARNING[10345]: loader.c:653 load_resource: Module 'chan_dadhi.so' could not be loaded. alderamin*CLI Mmmm... it would seem to be a bug: /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: No such file or directory Sounds like DAHDI was installed/compiled *after* Asterisk was compiled. Recompile Asterisk again and make sure /usr/lib/asterisk/modules/chan_dahdi.so is created when you make install. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Alert
Cary Fitch wrote: Can any one suggest a little code to either ring a cell phone when a new VM message is recorded, or send a text message? Basically outside sales people want to know they have a new message, but don't want to be interrupted to take a forwarded call. While a message by message notice would be nice, even just a single notice on the first message would be an alert to call for messages. Basically, a call from their own number would be the clue that there is a voicemail waiting. As someone else mentioned, if you want a text message I'd have asterisk send an email to an email-SMS gateway, you'll probably want to trim the email message down and prevent an attachment though. If you want an actual phone call, I have a series of scripts I use to call a person when a new voicemail is left. I use the vmnotify option in voicemail.conf to call a script that checks if the number of messages in the inbox is greater than the last time it was called (so a user doesn't get a call after they check their messages) and if so, create a call file to contact them and automatically connect them to the voice mail system. If you want it I can send it to you. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Polycom handset cord detangler
Hello list, I wondered if those of you using Polycom phones could recommend a decent cord detangler. I've had quite a few handsets get the tabs broken off in the jack from cord detanglers due to the recessed nature of the jack. This seems like it would work but I wanted some opinions before I go buy some: http://www.voiplink.com/Extended_Handset_Cord_Detangler_p/detangler-e.htm I personally hate detanglers. I get more complaints about static on calls that result from these things than anything else, but I need to provide some solution. Thanks -Dave P.S. For anyone looking for a way to repair damaged handsets with broken tabs, I've found that inserting the plug into the jack and then applying a blob of hot-glue in the notch will keep the cord secure in the handset. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
Jeff LaCoursiere wrote: Hmm, let me rephrase that (now that I have googled a bit). I am having trouble with DTMF tones over two IAX trunks: Polycom501---ast---[IAX]---ast2---[IAX]---provider Both IAX trunks were ulaw, and that worked fine. I recently changed the first leg to be g729 (as their internet connection is lower bandwidth). Now DTMF doesn't seem to pass. In my searches just now I see that dtmfmode is not actually a valid keyword in iax.conf. So may I assume that dtmfmode is inband only over IAX (since adding compression seems to have killed it?). That would suck. j It is my understanding that DTMF in IAX is *always* sent out of band. Make sure your Polycom and Asterisk are configured to use the same DTMF method in sip.conf. Polycom defaults to using rfc2833. It could have been prior to the G729 switch the DTMF audio just happened to be inband from the Polycom and Asterisk was configured to something different. Asterisk therefore didn't detect and translate the DTMF to out of band when it went over the IAX trunk. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sequential Ring Groups?
Marshall Henderson wrote:
Hi fellow Asterisk users!
I've got a PRI being used with a bunch of iaxmodems/Hylafax. I
currently have each individual channel of the PRI in its own context
that rings a specific iaxmodem. However, when a fax is complete on
that modem and another call comes into it, the modem is still in a
state of 'settling down' from the last call and I'd like to have it
ring a different channel if possible. So essentially, instead of each
PRI channel with a different context, I'd like to see one context that
simply handles calls to any available IAX peer (iaxmodem). How is this
done? I can certainly use the Dial() app to ring a bunch of extensions
at once but I'd like to have it try the first modem, if busy, then the
second, etc until one is available and answers.
I'm not using FreePBX for this fax server but am using it on my voice
PBX. Looking at the code for the 'firstavailable' ring group strategy
is of no help since its clouded within a whole mess of other functions
within [macro-dial]. Any ideas or pointers? THANKS!
This is how I'm doing it (AEL notation):
context inbound-pri {
FAXNBRHERE = {
Dial(IAX2/iaxmodem00/${EXTEN});
Dial(IAX2/iaxmodem01/${EXTEN});
Dial(IAX2/iaxmodem02/${EXTEN});
Dial(IAX2/iaxmodem03/${EXTEN});
// Et cetera ...
Busy();
}
}
Call comes in, starts at the top and if that modem is busy asterisk
moves on to the next line. If all the modems are busy then a busy
indication is sent back to the caller.
-Dave
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Re: [asterisk-users] DAHDI with OSLEC
Marco Sambo wrote: Mhmm. Thaht's strange! modinfo oslec -- modinfo: could not find module oslec and modinfo dahdi_echocan_oslec -- filename: /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko license:GPL author: Tzafrir Cohen tzafrir.co...@xorcom.com description:DAHDI OSLEC wrapper depends:dahdi vermagic: 2.6.26-1-486 mod_unload modversions 486 2009/3/31 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote: Hi, I've a problem: I can't configure DAHDI with ech canceller OSLEC. I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC. But when in /etc/dahdi/systems.conf I insert value echocanceller=oslec,1-4, command dahdi_cfg - give me an error about oslec. What is the output of: modinfo oslec modinfo dahdi_echocan_oslec -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ I'm not sure that is strange. When I build DAHDI with OSLEC I don't get an oslec module, I get an echo module: r...@srvpbx:~# modinfo echo filename: /lib/modules/2.6.27.19-smp/staging/echo/echo.ko version:0.3.0 description:Open Source Line Echo Canceller author: David Rowe license:GPL srcversion: 285EC80D84DCE294A677160 depends: vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 r...@srvpbx:~# modinfo dahdi_echocan_oslec filename: /lib/modules/2.6.27.19-smp/dahdi/dahdi_echocan_oslec.ko license:GPL author: Tzafrir Cohen tzafrir.co...@xorcom.com description:DAHDI OSLEC wrapper depends:dahdi,echo vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 Try building DAHDI with the steps detailed here and see if you have better luck: http://lists.digium.com/pipermail/asterisk-users/2009-January/225299.html -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI with OSLEC
Marco Sambo wrote: One thing! I saw that I use kernel 2.6.26 in my asterisk machine. I should use kernel 2.6.28 or newer to use oslec with DAHDI??? You don't need to, if you read me previous email you'll notice I'm running 2.6.27.19. Rebuild DAHDI with the instructions I linked to and you'll get the echo module with DADHI. It requires you download 2.6.28 but not that you are running 2.6.28. 2009/4/1 Marco Sambo derwid...@gmail.com But I don't have also echo modinfo echo modinfo: could not find module echo 2009/4/1 Dave Fullerton dfullertaster...@shorelinecontainer.com Marco Sambo wrote: Mhmm. Thaht's strange! modinfo oslec -- modinfo: could not find module oslec and modinfo dahdi_echocan_oslec -- filename: /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko license:GPL author: Tzafrir Cohen tzafrir.co...@xorcom.com description:DAHDI OSLEC wrapper depends:dahdi vermagic: 2.6.26-1-486 mod_unload modversions 486 2009/3/31 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote: Hi, I've a problem: I can't configure DAHDI with ech canceller OSLEC. I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC. But when in /etc/dahdi/systems.conf I insert value echocanceller=oslec,1-4, command dahdi_cfg - give me an error about oslec. What is the output of: modinfo oslec modinfo dahdi_echocan_oslec -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com jabber%3atzafrir.co...@xorcom.com jabber%253atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ I'm not sure that is strange. When I build DAHDI with OSLEC I don't get an oslec module, I get an echo module: r...@srvpbx:~# modinfo echo filename: /lib/modules/2.6.27.19-smp/staging/echo/echo.ko version:0.3.0 description:Open Source Line Echo Canceller author: David Rowe license:GPL srcversion: 285EC80D84DCE294A677160 depends: vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 r...@srvpbx:~# modinfo dahdi_echocan_oslec filename: /lib/modules/2.6.27.19-smp/dahdi/dahdi_echocan_oslec.ko license:GPL author: Tzafrir Cohen tzafrir.co...@xorcom.com description:DAHDI OSLEC wrapper depends:dahdi,echo vermagic: 2.6.27.19-smp SMP preempt mod_unload 686 Try building DAHDI with the steps detailed here and see if you have better luck: http://lists.digium.com/pipermail/asterisk-users/2009-January/225299.html -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- DO NOT SEND WITH THIS ACCOUNT ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATT PRI Install - What is outpulsed?
Hey All, ATT is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have any idea what she was referring to? Is this ANI? Outgoing Caller ID? Something else? While I've done many POTS line setups, this is my first PRI install, so I'd also welcome any make sure you do this, read this first or ATT always messes this up so... tips. Thanks -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Controlling BLF Leds ...
Gordon Henderson wrote: Is there a way to set/clear a BLF LED on a phone from the dialplan? I want to use one as an indicator of some state in the PBX - in this case it's night mode but I can think of other applications. I have BLFs working just fine for normal stuff, just wonderin if I can use them for more! Cheers, Gordon I think this is what you want: http://www.voip-info.org/wiki/view/Asterisk+func+Devstate -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kewlstart - Busy signal before battery drop.
Hello all. I have Asterisk connected to an Adit 600 channel bank with a TE110P and the channel bank is connected to a PBX providing dialtone to the PBX with fxo_ks signalling. When a call between the PBX and Asterisk completes there is a momentary battery drop/reversal or something that signals the PBX that the other side has hung up and then the PBX hangs up. This all works fine. However, when asterisk hangs up it also immediately starts playing a busy signal. My issue is that the busy signal begins playing before the battery drop occurs. This means that at the end of any calls or voicemails on the PBX there is a .5-1 second interval of a busy tone at the end. Is there any way to get the busy tone to begin *after* the battery drop? I've tried messing with the indications.conf file but didn't have any luck and I can't see anything in chan_dahdi.conf or system.conf. This same thing happens at home with my TDM400P so I'm inclined to think it's not exclusive to the channel bank. Anyone have any ideas? Thanks -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with incoming and outgoing calls via TDM
Rosa De Santis wrote: Hello all. Please, I'd like to know if somebody can help me with this problem. I have successfully configured a PBX with Asterisk 1.4 and a Digium analog card with 4 ports. This PBX has a lot of incoming and outgoing calls, and works perfect in general, but there are some extrange cases where an incoming call is bridget with an outgoing call, and the caller that is calling TO the PBX can even hear the dtmf tones of the caller that is calling OUT the PBX, and due the high traffic this is happening a lot. It seems that asterisk is taking the zap channel to call out in the exact moment before it is marked as busy with the incoming call. Please, is there any configuration to avoid this? Thanks a lot in advance. Rosa. The situation you're referring to is called glare. You'll find discussion of it in the archives and on voip-info.org. You need to make sure you are seizing lines for outgoing calls in the reverse order that they are used for incoming calls. Check out the G dialing option for Zaptel/DAHDI channels (under Dialing a Group section): http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels If this doesn't work, your next best bet is to increase the number of lines you have. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] configuring channels for dahdi
Aqua Man wrote: after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI module load chan_dahdi.so receive the following: signalling must be specified before any channels are. CLI Warning [4663]: chan_dahdi.c:11627 process_dahdi: Ignoring signalling Error[4663]: chan_dahdi.c:10946 build_channels: Unable to reconfigure channel '1' Error[4663]: chan_dahdi.c:11970 reload: Reload of chan_dahdi.so is unsuccessful! NOTICE[4641]: loader.c:580 ast_module_reload: The module 'chan_dahdi.so' was not properly initialized. Before reloading the module, you must run 'module load chan_dahdi.so' and fix whatever is preventing the module from being initialized. dahdi_cfg -vvv dahdi version: 2.1.0.4 Echo Canceller(s): mg2 Configuration Channel map: channel 01: fxo kewlstart (Default) (Echo Canceler: mg2) (Slaves:01) channel 04: fxs kewlstart (Default) (Echo Canceler: mg2) (Slaves:04) snip What are the contents of chan_dahdi.conf in /etc/asterisk? Did you specify what signalling to use there? -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk analog DID with Adit 600
Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed to DTMF) signalling. What I've done is purchase an Adit 600 with an FXO card. I've set the Adit T1 controller to use EM signalling and the FXO card to use DPT signalling. I've set asterisk to use EM in both dahdi/system.conf and chan_dahdi.conf. If I dial the port connected to the DID card it goes off hook but when the Executone winks Asterisk or the Adit thinks the remote side has hung up and terminates the call. I've tried using EM wink, featd, and featb signalling in chan_dahdi.conf, set hanguponpolarityswitch=no, and tried loop start signalling for the heck of it and that didn't work either. Does anyone have any suggestions of additional things I could try? Thanks in advance, -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SOLVED - Re: Asterisk analog DID with Adit 600
Dave Fullerton wrote: Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed to DTMF) signalling. What I've done is purchase an Adit 600 with an FXO card. I've set the Adit T1 controller to use EM signalling and the FXO card to use DPT signalling. I've set asterisk to use EM in both dahdi/system.conf and chan_dahdi.conf. If I dial the port connected to the DID card it goes off hook but when the Executone winks Asterisk or the Adit thinks the remote side has hung up and terminates the call. I've tried using EM wink, featd, and featb signalling in chan_dahdi.conf, set hanguponpolarityswitch=no, and tried loop start signalling for the heck of it and that didn't work either. Does anyone have any suggestions of additional things I could try? Thanks in advance, -Dave Please disregard. Everything works much better when the wires are connected to the correct pair. I had the tip of pair A and the ring from pair B. Thanks anyways. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Phones start to break up after beingup a LONG time
Jeff LaCoursiere wrote: On Fri, 20 Feb 2009, Danny Nicholas wrote: This is just a hack, but why don't you schedule a sip notify polycom-restart during lunch hour? You could run it from a cron job using this line for each phone: Asterisk -rx sip notify polycom-check-cfg 100 replacing 100 with the number of the phone (extension). Hey this would be neat! But I cannot get it to work: Connected to Asterisk 1.4.23.1 currently running on pbx (pid = 15728) Verbosity was 0 and is now 1 pbx*CLI sip notify polycom-check-cfg 223 Unable to find notify type 'polycom-check-cfg' pbx*CLI Must it be defined somewhere? Cheers, Yes, you need the sip_notify.conf file in /etc/asterisk. The sample file that's in the asterisk source has the definition for polycom's in it. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDMOE Timing
Hello all, I have two machines I'm connecting with TDMOE (dahdi dynamic spans) and I have a question about timing parameters. By my understanding one machine should be the source of the timing and the other a slave of that timing. So on machine A I have the following in system.conf: dynamic=eth,eth0/00:0C:29:55:89:7E,24,0 On machine B I have this is system.conf: dynamic=eth,eth0/00:18:8B:C7:F6:94,24,1 So machine A is the source of timing and B is slave to it. If both of these machines also have a digium (TDM400P in one and a TE110P in the other) card in them is this configuration still correct or should I use 0 for timing on both? The reason I ask is if I boot both machines fresh and I execute dahdi_cfg on machine A first and then machine B I either get a kernel oops (with 2.6.27.11) or complete freeze (with 2.6.23.17) on machine B pretty much without fail. If I do machine B first and then A everything works fine. I'm using dahdi_linux 2.1.0.4 on both. I know I can just use SIP or IAX or anything else to connect these two machines, but I'm using this as a learning experience to play with PRI setups. Thanks -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Test Lab
Lee Wilson wrote: Hey Everyone, I would like to start testing/playing with PRI channels but I don't have access to a PRI line. Is it possible to do the equivilent of a crossover between two PRI Cards (say Digium's TE120P)? What I was thinking is that I could set one asterisk box up with a PRI card set as the TE and provide clocking and another box exactly the same but with the card setup as NT. I think I would also need to wire up the correct type of crossover as a standard ethernet crossover would not work or would it? Thanks in advance. Lee Since you have gotten plenty of responses on this I thought I'd throw out another option. If you just want to play with how a PRI connection behaves without all the hardware investment, you can emulate a PRI over TDMOE. I did it a while back just to see how calls were passed back and forth and how result codes were set. Everything is configured the same in asterisk, you just use a dynamic span instead of a physical one. You will still need one side to have a timing source (I did get mine to work with just ztdummy). -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you use? .conf or AEL?
Alan Lord (News) wrote: Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed new and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA I use AEL. I find it much cleaner to look at and not having to deal with priorities is a bonus. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oslec + dahdi
Tzafrir Cohen wrote: On Wed, Jan 21, 2009 at 06:35:58PM -0600, troxlinux wrote: Hi list, I install dahdi-linux successfully with the module of oslec for the echo, but when I specify it in the system.conf the echo canceller oslec it shows me errors: DAHDI_ATTACH_ECHOCAN failed on channel 4: Invalid argument (22) What version have you installed? This sounds similar to a post on the OSLEC mailing list (no resolution there either): http://sourceforge.net/mailarchive/forum.php?thread_name=Pine.OSX.4.64.0901121456390.25971%40john.brc.ubc.caforum_name=freetel-oslec I'm having the same issue with dahdi-linux-2.1.0.3 using the staging drivers from 2.6.28. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oslec + dahdi
Vincent Li wrote: On Thu, 22 Jan 2009, troxlinux wrote: I have dahdi-linux-2.1.0.3 in centos 5.2 and the last version oslec svn I have installed oslec and loaded, but it doesn't work me with dahdi modinfo oslec filename: /lib/modules/2.6.18-92.1.22.el5/kernel/net/ipv4/oslec.ko description:Open Source Line Echo Canceller Zaptel Wrapper author: David Rowe license:GPL srcversion: 13813ACD4A228F69FF4B5C1 depends: vermagic: 2.6.18-92.1.22.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4. oslec is a great great great software, with the version of zaptel 1.4.11 I had it installed and without anything of echo in my card TDM 400 I almost have the same enviroment as you, I basically run the following script to get oslec work with my tdm411 card. #!/bin/sh cd /usr/src wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2 tar xjf linux-2.6.28.tar.bz2 wget http://downloads.digium.com/pub/telephony/dahdi-tools/dahdi-tools-2.1.0.2.tar.gz wget http://downloads.digium.com/pub/telephony/dahdi-linux/dahdi-linux-2.1.0.3.tar.gz tar zxvf dahdi-linux-2.1.0.3.tar.gz ln -s /usr/src/dahdi-linux-2.1.0.3 /usr/src/dahdi mkdir /usr/src/dahdi/drivers/staging cp -fR /usr/src/linux-2.6.28/drivers/staging/echo /usr/src/dahdi/drivers/staging sed -i s|#obj-m += dahdi_echocan_oslec.o|obj-m += dahdi_echocan_oslec.o| /usr/src/dahdi/drivers/dahdi/Kbuild sed -i s|#obj-m += ../staging/echo/|obj-m += ../staging/echo/| /usr/src/dahdi/drivers/dahdi/Kbuild echo 'obj-m += echo.o' /usr/src/dahdi/drivers/staging/echo/Kbuild cd /usr/src/dahdi make make install cd /usr/src tar zxvf dahdi-tools-2.1.0.2.tar.gz cd /usr/src/dahdi-tools-2.1.0.2 ./configure make make install Hope it helps. Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' Thanks! I think this is the part we were missing: echo 'obj-m += echo.o' /usr/src/dahdi/drivers/staging/echo/Kbuild Any chance someone could add that line into the dahdi-linux README? It now modprobe's without issues. I'll get to trying it out later. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Executive Assistant Guidance
Jeremy Mann wrote: Looking for two things: 1. Anyone that has dialplan logic for an executive assistant. My owners want their extensions to ring on her phone, and be very obvious to her which extension is ringing. They also want her to have presense. She's got Polycom IP 650 with sidecar, they have IP 550 phones. Thusfar I've got her registering to 4 extensions. Each extension is labeled with an executive and rings alongside theirs(Dial(SIP/126SIP/191)) just didn't know if there was a better way. I also have presense setup on her Sidecar but it only has one status, is there a way for her to know their line is ringing and not just in use. ? 2. Sort of tied to #1, does anyone have clear dialplan logic and polycom config information about doing custom ringing per extension on the IP 650 ? You have two options for #2: You can define a SIP alertInfo for different rings in the sip.cfg and then in asterisk set the alert info header in the dialplan. You can change the reg.x.ringType on each registration in the phone's config file. See the SIP admin guide for details. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use AMD Answering Machine Detect ?
Daniel Varella wrote:
Hi everybody,
Happy New Year !
I'm trying to detect if a call was answered by a machine (linke
voicemail systems) or a human.
I would like to use AMD (Answering Machine Detect) command, but
with my configuration it was not possible get there.
Follow my dialplan:
exten = _[789].,1,NoCDR
exten = _[789].,n,Dial(SIP/${ext...@111,60)
exten = _[789].,n,AMD
exten = _[789].,n,NoOp(AMD Status is: ${AMDSTATUS})
exten = _[789].,n,Hangup
What is happening is when the call is answered by the other part,
Asterisk doesn't go to the next level (exten = _[789].,n,AMD). So AMD
can't verify the call.
How can I do this ? Any idea ?
Thanks in advance.
The Dial app will not exit until the call is completed (one or both
parties hang up). You need to put what you want to happen during the
call in a macro and then call the macro with the M() option to Dial (see
the Dial app help text).
-Dave
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Re: [asterisk-users] [Asus Eeebox] USB FXO adapter?
Vincent wrote: Hello I'm contemplating building an Asterisk voice server out of the compact Asus EeeBox: http://www.asus.com/products.aspx?l1=24l2=165 But they're so compact, they don't have a PCI slot to handle an analog phone line. I'd like to minimize footpring and cables: Besides analog/SIP boxes like Linksys (extra cables + transformer), does someone know of a USB adapter that is self-powered and could take an analog line as input, convert voice to SIP, and send packets through the USB port? Thank you. It hasn't been released yet, but this looks like it will do the job: http://wiki.sangoma.com/sangoma-wanpipe-usbfxo People have been reviewing betas since early September so hopefully it will be released soon. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: Cannot conference with 10 digit 3rd party
Lee, John (Sydney) wrote: Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. dialplan dialplan.impossibleMatchHandling=2 /dialplan (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size telephone number and then hit the send or dial key. This works quite well except when I am doing conferencing. It goes like this: I dialled the 1st party and was answered. Then I press conf key and then enter the 3rd party. I can keep entering until it reaches the 10th digit and then the 10-digit number is automatically dialled. Any thoughts? I don't think the 2 works quite that way. From what I read in the admin guide the impossibleMatchHandling lets you tell the phone how it should handle numbers that are dialed that do NOT match the dial plan. Your numbers that are longer than 10 digits probably match one of the entries in the phone's dialplan so as soon as it matches it sends the number to asterisk. You will either need to wipe out the phone dialplan and replace it with a generic X.T or add a digit map for the number you are dialing that is greater than 10 digits long. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL: how to check if variable is defined
Philipp Kempgen wrote:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
I use an if condition in extensions.ael to check if a channel variable
is defined and if defined I add a certain header:
context toNormaleRufe {
_X. = {
if (${NUMBER}) {
SIPAddHeader(X-NUMBER: ${NUMBER});
};
...
};
This works fine, except NUMBER starts with the + sign.
I tried using quotes but
if (${NUMBER})
evaluates always true.
What is the suggested way to solve this?
if (${NUMBER} != ) {
// ...
}
That doesn't tell you whether the variable is defined but in
most cases (if any) that doesn't matter anyway.
But I guess it wouldn't hurt to add a DEFINED() function to
Asterisk.
if (DEFINED(myvariable)) {
// ...
}
Isn't that what EXISTS() is for?
-Dave
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Re: [asterisk-users] AEL Variable Warning Messages
Brent Davidson wrote:
Philipp Kempgen wrote:
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression
Zap/r2 has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item:
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression
Zap/r1 has operators, but no variables. Interesting...
I'd suggest
Set(TRUNK=Zap/r2);
resp.
Set(TRUNK=Zap/r1);
Philipp Kempgen
According to the AEL Documentation I should be able to set variables
without using the Set command. They even give the following example:
context foo {
555 = {
x=5;
y=blah;
divexample=10/2
NoOp(x is ${x} and y is ${y} !);
};
};
I wonder if maybe AEL is ignoring the double quotes and treating the
Zap/r2 as if it were division??? Should I file a bug report on this?
I had gotten similar messages when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk are you running?
-Dave
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Re: [asterisk-users] DAHDI install dont need download of echo cancel
Mr. James W. Laferriere wrote: Hello Tzafir , On Thu, 18 Dec 2008, Tzafrir Cohen wrote: On Thu, Dec 18, 2008 at 12:48:59PM -0500, Matt Watson wrote: after you have configured zaptel manually the first time, copy the menuselect.makeopts file that is generated in the root directory of the zaptel source to a file /etc/zaptel.makeopts. presumably this is available for people that have moved on to DAHDI as well, and I would guess it should be /etc/dahdi.makeopts - but I have not verified that. dahdi-linux does not use menuselect. Then can someone tell me why this file exists ? /home/archive/asterisk/dahdi-linux-complete-2.0.0+2.0.0/tools/menuselect.makeopts # cat !$ MENUSELECT_UTILS=fxstest sethdlc dahdi_diag dahdi_tool MENUSELECT_BUILD_DEPS= MENUSELECT_DEPSFAILED=MENUSELECT_UTILS=sethdlc MENUSELECT_DEPSFAILED=MENUSELECT_UTILS=dahdi_tool Tia , JimL You're using the combined tarball that has both dahdi-linux and dahdi-tools. That makeopts files is for the tools side (as shown the path). -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk v1.6 on Ubuntu Intrepid?
Christian wrote: Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c - manager.o manager.c: In function ‘action_getvar’: manager.c:1732: error: ‘SENTINEL’ undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 Are you by chance using 1.6.0.2? Try grabbing 1.6.0.3-rc1 or 1.6.0.1 instead. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
Brent Davidson wrote: I have several branch offices all running Asterisk PBX's that register to each other via SIP so that calls can be transferred from office to office. Everything is working great on the office to office transfers, but I'd like to somehow make the CallerID more useful. Currently if an extension at Office1 dials an extension at Office2 the CID on the phone at Office2 says Office1. The same thing happens if a person at Office1 transfers an incoming call to Office2. The caller ID at Office2 always just says Office1. What I would like to happen would be when Bob at Extension 12 at Office1 calls Office2 the caller ID at office 2 would say Bob in the name files and 12 in the number field. If Bob does a blind transfer to an extension at Office2 I would like the caller ID on the Office2 phone to display the original caller's name and number. I've read most of the documentation on the CallerID variables, but am still having a bit of trouble wrapping my head around the necessary logic to accomplish what I need to do, (mainly because I'm in the middle of a totally unrelated project and am having trouble multi-tasking). Could anyone give me a starting point? Thanks, Brent Check the entries for office1 and office2 servers in sip.conf. If they have a callerid= entry comment it out and do a SIP reload. When it is set asterisk overrides the caller ID sent to it. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
I'm not overriding the callerID anywhere in my dial plan.
Would the way I route the extensions make any difference? Each office
has it's own server and prefix by which it is accessed from another
office. So for office1 to dial extension 12 at office2 he would dial 1012.
In my Dialplan I have (AEL syntax):
_10XX = {
Dial(SIP/${EXTEN:2...@office2,,Tt);
Hangup;
}
And in my SIP.conf on Office 1
[Office2]
username=Office1-user
fromuser=Office1-user
host=XXX.XXX.XXX.XXX (edited out)
type=peer
context=internal
secret= password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
qualify=yes
canreinvite=no
In My Sip.Conf on Office2:
[Office1-user]
username=Office1
host=XXX.XXX.XXX.XXX (edited out)
type=user
context=internal
secret=password
dtmfmode=rfc2833
disallow=all
allow=speex
call-limit=20
canreinvite=no
Separating into peer and user entries was the only way I was able to get
calls to go through and be authenticated properly. Would this setup
have any bearing on the caller ID?
I don't see anything sticking out as being wrong. For kicks, what is the
output of sip show user Office1-user on office2?
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Re: [asterisk-users] SIP CallerID Question
Brent Davidson wrote:
Dave Fullerton wrote:
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If
they have a callerid= entry comment it out and do a SIP reload. When
it is set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
I'm not overriding the callerID anywhere in my dial plan.
Would the way I route the extensions make any difference? Each
office has it's own server and prefix by which it is accessed from
another office. So for office1 to dial extension 12 at office2 he
would dial 1012.
In my Dialplan I have (AEL syntax):
_10XX = {
Dial(SIP/${EXTEN:2...@office2,,Tt);
Hangup;
}
I don't see anything sticking out as being wrong. For kicks, what is
the output of sip show user Office1-user on office2?
___
localhost*CLI sip show user Office1-user
localhost*CLI
* Name : Office1-user
Secret : Set
MD5Secret: Not set
Context : internal
Language : en
AMA flags: Unknown
Transfer mode: open
MaxCallBR: 384 kbps
CallingPres : Presentation Allowed, Not Screened
Call limit : 20
Callgroup:
Pickupgroup :
Callerid :
ACL : No
Codec Order : (speex:20)
Auto-Framing: No
If user A in office1 calls user B in office1 does caller ID work then?
If yes, then I'm afraid I'm out of ideas. If no, then make sure the
extensions have caller id set either in sip.conf or by the phone itself.
-Dave
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Re: [asterisk-users] Asterisk variable for SIP context
Mike wrote: Hi, Say I wanted to know what context a SIP registration is using to dial out in my dialplan, what would I do? For example, I have phones on a local-calls-only context (as defined in sip.conf), others in unrestricted-calls. In my dialplan, I`d like to act on that knowledge. Mike The SIPPEER function should allow you to extract what context is defined in sip.conf. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging, Polycom and whispers
Mike wrote: Hi, Is there a way to page a Polycom phone that is already in use (if, of course, the call isn't on speakerphone already)? I've never been able to find a way. Any attempt I made either put the existing call on hold to auto-answer the page or the page just rang at the phone and then caused other issues. I'm not sure you'll have any luck with other SIP phones either. What you're asking it to do is accept two simultaneous calls but put each call on a different listening device (handset/speakerphone in this case). The closest you might get is to rig a dialplan that would use chanspy in whisper mode to play the page through the current audio device if the phone is busy. I don't know how to go about doing that however. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 I see a reference in the 1.6 changelog that refers to SENTINEL not existing in 1.6.0 2008-06-27 01:09 + [r125648-125684] Mark Michelson [EMAIL PROTECTED] * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
Tzafrir Cohen wrote: On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote: Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 On what platform is it? Slackware 12.0 1.6.0.1 compiles fine. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
Olivier wrote: Hello, Groups in asterisk are summarized here ( http://www.voip-info.org/wiki/view/Channels+and+Groups). Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789 (as I've been advised in another thread, to switch from one notation to the other and I can't see the reason behind that) ? Regards Assuming nothing has changed from Zaptel to DAHDI, the difference can found here: http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels Basically, the lowercase g chooses the lowest number available channel from the group where the uppercase G chooses the highest number available channel. This is used to reduce glare on analog or T1 (non-PRI) channels that are part of a hunt group. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX2 with delayed echo
There are also settings which will turn on local echo cancellation for the handset, headset and/or speaker phone. I don't recall their names at the moment. They are off by default on the handset and headset unless you're using a very recent (3.0+) SIP app. Tim Nelson wrote: I'm not sure about the 3 second delay, but I've seen plenty of echo issues on Polycom phones when the gain has been changed on the handset. Check the voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're not too high. You also may want to make sure there aren't any system resource constraints such as high CPU usage or memory usage... :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - c james [EMAIL PROTECTED] wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what they wanted to use!) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
